navigator.getUserMedia 不是使用 WebRTC 的函数
navigator.getUserMedia is not a function using WebRTC
我尝试与一些人建立一个视频组会议流，我使用 WebRTC（来自这里：https://github.com/anoek/webrtc-group-chat-example），作为服务器，我使用 VPS 安装了节点 js 的 Centos 7，当我尝试启动我的 webrtc 服务器，在浏览器中，我在使用端口
访问 ip 时看到此错误
这个错误：
(index):260 Uncaught TypeError: navigator.getUserMedia is not a function
at setup_local_media ((index):260)
at Socket.<anonymous> ((index):50)
at Socket.Emitter.emit (socket.io-1.4.5.js:3)
at Socket.emit (socket.io-1.4.5.js:3)
at Socket.onconnect (socket.io-1.4.5.js:3)
at Socket.onpacket (socket.io-1.4.5.js:3)
at Manager.<anonymous> (socket.io-1.4.5.js:3)
at Manager.Emitter.emit (socket.io-1.4.5.js:3)
at Manager.ondecoded (socket.io-1.4.5.js:2)
at Decoder.<anonymous> (socket.io-1.4.5.js:3)
JS代码：
/*************/
/*** SETUP ***/
/*************/
var express = require('express');
var bodyParser = require('body-parser')
var cors = require('cors')
var main = express()
main.use(cors())
var server = main.listen(8000);
var io = require('socket.io').listen(server);
main.get('/', function(req, res){ res.sendFile(__dirname + '/client.html'); });
/*************************/
/*** INTERESTING STUFF ***/
/*************************/
var channels = {};
var sockets = {};
/**
* Users will connect to the signaling server, after which they'll issue a "join"
* to join a particular channel. The signaling server keeps track of all sockets
* who are in a channel, and on join will send out 'addPeer' events to each pair
* of users in a channel. When clients receive the 'addPeer' even they'll begin
* setting up an RTCPeerConnection with one another. During this process they'll
* need to relay ICECandidate information to one another, as well as SessionDescription
* information. After all of that happens, they'll finally be able to complete
* the peer connection and will be streaming audio/video between eachother.
*/
io.sockets.on('connection', function (socket) {
socket.channels = {};
sockets[socket.id] = socket;
console.log("["+ socket.id + "] connection accepted");
socket.on('disconnect', function () {
for (var channel in socket.channels) {
part(channel);
}
console.log("["+ socket.id + "] disconnected");
delete sockets[socket.id];
});
socket.on('join', function (config) {
console.log("["+ socket.id + "] join ", config);
var channel = config.channel;
var userdata = config.userdata;
if (channel in socket.channels) {
console.log("["+ socket.id + "] ERROR: already joined ", channel);
return;
}
if (!(channel in channels)) {
channels[channel] = {};
}
for (id in channels[channel]) {
channels[channel][id].emit('addPeer', {'peer_id': socket.id, 'should_create_offer': false});
socket.emit('addPeer', {'peer_id': id, 'should_create_offer': true});
}
channels[channel][socket.id] = socket;
socket.channels[channel] = channel;
});
function part(channel) {
console.log("["+ socket.id + "] part ");
if (!(channel in socket.channels)) {
console.log("["+ socket.id + "] ERROR: not in ", channel);
return;
}
delete socket.channels[channel];
delete channels[channel][socket.id];
for (id in channels[channel]) {
channels[channel][id].emit('removePeer', {'peer_id': socket.id});
socket.emit('removePeer', {'peer_id': id});
}
}
socket.on('part', part);
socket.on('relayICECandidate', function(config) {
var peer_id = config.peer_id;
var ice_candidate = config.ice_candidate;
console.log("["+ socket.id + "] relaying ICE candidate to [" + peer_id + "] ", ice_candidate);
if (peer_id in sockets) {
sockets[peer_id].emit('iceCandidate', {'peer_id': socket.id, 'ice_candidate': ice_candidate});
}
});
socket.on('relaySessionDescription', function(config) {
var peer_id = config.peer_id;
var session_description = config.session_description;
console.log("["+ socket.id + "] relaying session description to [" + peer_id + "] ", session_description);
if (peer_id in sockets) {
sockets[peer_id].emit('sessionDescription', {'peer_id': socket.id, 'session_description': session_description});
}
});
});
Html代码：
<!doctype html>
<html>
<head>
<script src="http://ajax.googleapis.com/ajax/libs/jquery/1.10.2/jquery.min.js"></script>
<!-- This adapter.js file de-prefixes the webkit* and moz* prefixed RTC
methods. When RTC becomes a more solid standard, this adapter should no
longer be necessary. -->
<!-- <script src="https://webrtc.googlecode.com/svn/trunk/samples/js/base/adapter.js"></script> -->
<style>
html, body {
background-color: #333;
}
video {
width: 320px;
height: 240px;
border: 1px solid black;
}
</style>
<script src="https://cdn.socket.io/socket.io-1.4.5.js"></script>
<script>
/** CONFIG **/
var SIGNALING_SERVER = "http://127.0.0.1:8000";
var USE_AUDIO = true;
var USE_VIDEO = true;
var DEFAULT_CHANNEL = 'some-global-channel-name';
var MUTE_AUDIO_BY_DEFAULT = false;
/** You should probably use a different stun server doing commercial stuff **/
/** Also see: https://gist.github.com/zziuni/3741933 **/
var ICE_SERVERS = [
{url:"stun:stun.l.google.com:19302"}
];
</script>
<script>
var signaling_socket = null; /* our socket.io connection to our webserver */
var local_media_stream = null; /* our own microphone / webcam */
var peers = {}; /* keep track of our peer connections, indexed by peer_id (aka socket.io id) */
var peer_media_elements = {}; /* keep track of our <video>/<audio> tags, indexed by peer_id */
function init() {
console.log("Connecting to signaling server");
signaling_socket = io(SIGNALING_SERVER);
signaling_socket = io();
signaling_socket.on('connect', function() {
console.log("Connected to signaling server");
setup_local_media(function() {
/* once the user has given us access to their
* microphone/camcorder, join the channel and start peering up */
join_chat_channel(DEFAULT_CHANNEL, {'whatever-you-want-here': 'stuff'});
});
});
signaling_socket.on('disconnect', function() {
console.log("Disconnected from signaling server");
/* Tear down all of our peer connections and remove all the
* media divs when we disconnect */
for (peer_id in peer_media_elements) {
peer_media_elements[peer_id].remove();
}
for (peer_id in peers) {
peers[peer_id].close();
}
peers = {};
peer_media_elements = {};
});
function join_chat_channel(channel, userdata) {
signaling_socket.emit('join', {"channel": channel, "userdata": userdata});
}
function part_chat_channel(channel) {
signaling_socket.emit('part', channel);
}
/**
* When we join a group, our signaling server will send out 'addPeer' events to each pair
* of users in the group (creating a fully-connected graph of users, ie if there are 6 people
* in the channel you will connect directly to the other 5, so there will be a total of 15
* connections in the network).
*/
signaling_socket.on('addPeer', function(config) {
console.log('Signaling server said to add peer:', config);
var peer_id = config.peer_id;
if (peer_id in peers) {
/* This could happen if the user joins multiple channels where the other peer is also in. */
console.log("Already connected to peer ", peer_id);
return;
}
var peer_connection = new RTCPeerConnection(
{"iceServers": ICE_SERVERS},
{"optional": [{"DtlsSrtpKeyAgreement": true}]} /* this will no longer be needed by chrome
* eventually (supposedly), but is necessary
* for now to get firefox to talk to chrome */
);
peers[peer_id] = peer_connection;
peer_connection.onicecandidate = function(event) {
if (event.candidate) {
signaling_socket.emit('relayICECandidate', {
'peer_id': peer_id,
'ice_candidate': {
'sdpMLineIndex': event.candidate.sdpMLineIndex,
'candidate': event.candidate.candidate
}
});
}
}
peer_connection.onaddstream = function(event) {
console.log("onAddStream", event);
var remote_media = USE_VIDEO ? $("<video>") : $("<audio>");
remote_media.attr("autoplay", "autoplay");
if (MUTE_AUDIO_BY_DEFAULT) {
remote_media.attr("muted", "true");
}
remote_media.attr("controls", "");
peer_media_elements[peer_id] = remote_media;
$('body').append(remote_media);
attachMediaStream(remote_media[0], event.stream);
}
/* Add our local stream */
peer_connection.addStream(local_media_stream);
/* Only one side of the peer connection should create the
* offer, the signaling server picks one to be the offerer.
* The other user will get a 'sessionDescription' event and will
* create an offer, then send back an answer 'sessionDescription' to us
*/
if (config.should_create_offer) {
console.log("Creating RTC offer to ", peer_id);
peer_connection.createOffer(
function (local_description) {
console.log("Local offer description is: ", local_description);
peer_connection.setLocalDescription(local_description,
function() {
signaling_socket.emit('relaySessionDescription',
{'peer_id': peer_id, 'session_description': local_description});
console.log("Offer setLocalDescription succeeded");
},
function() { Alert("Offer setLocalDescription failed!"); }
);
},
function (error) {
console.log("Error sending offer: ", error);
});
}
});
/**
* Peers exchange session descriptions which contains information
* about their audio / video settings and that sort of stuff. First
* the 'offerer' sends a description to the 'answerer' (with type
* "offer"), then the answerer sends one back (with type "answer").
*/
signaling_socket.on('sessionDescription', function(config) {
console.log('Remote description received: ', config);
var peer_id = config.peer_id;
var peer = peers[peer_id];
var remote_description = config.session_description;
console.log(config.session_description);
var desc = new RTCSessionDescription(remote_description);
var stuff = peer.setRemoteDescription(desc,
function() {
console.log("setRemoteDescription succeeded");
if (remote_description.type == "offer") {
console.log("Creating answer");
peer.createAnswer(
function(local_description) {
console.log("Answer description is: ", local_description);
peer.setLocalDescription(local_description,
function() {
signaling_socket.emit('relaySessionDescription',
{'peer_id': peer_id, 'session_description': local_description});
console.log("Answer setLocalDescription succeeded");
},
function() { Alert("Answer setLocalDescription failed!"); }
);
},
function(error) {
console.log("Error creating answer: ", error);
console.log(peer);
});
}
},
function(error) {
console.log("setRemoteDescription error: ", error);
}
);
console.log("Description Object: ", desc);
});
/**
* The offerer will send a number of ICE Candidate blobs to the answerer so they
* can begin trying to find the best path to one another on the net.
*/
signaling_socket.on('iceCandidate', function(config) {
var peer = peers[config.peer_id];
var ice_candidate = config.ice_candidate;
peer.addIceCandidate(new RTCIceCandidate(ice_candidate));
});
/**
* When a user leaves a channel (or is disconnected from the
* signaling server) everyone will recieve a 'removePeer' message
* telling them to trash the media channels they have open for those
* that peer. If it was this client that left a channel, they'll also
* receive the removePeers. If this client was disconnected, they
* wont receive removePeers, but rather the
* signaling_socket.on('disconnect') code will kick in and tear down
* all the peer sessions.
*/
signaling_socket.on('removePeer', function(config) {
console.log('Signaling server said to remove peer:', config);
var peer_id = config.peer_id;
if (peer_id in peer_media_elements) {
peer_media_elements[peer_id].remove();
}
if (peer_id in peers) {
peers[peer_id].close();
}
delete peers[peer_id];
delete peer_media_elements[config.peer_id];
});
}
/***********************/
/** Local media stuff **/
/***********************/
function setup_local_media(callback, errorback) {
if (local_media_stream != null) { /* ie, if we've already been initialized */
if (callback) callback();
return;
}
/* Ask user for permission to use the computers microphone and/or camera,
* attach it to an <audio> or <video> tag if they give us access. */
console.log("Requesting access to local audio / video inputs");
navigator.getUserMedia = ( navigator.getUserMedia ||
navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia ||
navigator.msGetUserMedia);
attachMediaStream = function(element, stream) {
console.log('DEPRECATED, attachMediaStream will soon be removed.');
element.srcObject = stream;
};
navigator.getUserMedia({"audio":USE_AUDIO, "video":USE_VIDEO},
function(stream) { /* user accepted access to a/v */
console.log("Access granted to audio/video");
local_media_stream = stream;
var local_media = USE_VIDEO ? $("<video>") : $("<audio>");
local_media.attr("autoplay", "autoplay");
local_media.attr("muted", "true"); /* always mute ourselves by default */
local_media.attr("controls", "");
$('body').append(local_media);
attachMediaStream(local_media[0], stream);
if (callback) callback();
},
function() { /* user denied access to a/v */
console.log("Access denied for audio/video");
alert("You chose not to provide access to the camera/microphone, demo will not work.");
if (errorback) errorback();
});
}
</script>
</head>
<body onload='init()'>
<!--
the <video> and <audio> tags are all added and removed dynamically
in 'onAddStream', 'setup_local_media', and 'removePeer'/'disconnect'
-->
</body>
</html>
navigator.getUserMedia 仅在安全上下文（即 https 或本地主机）中可用 - 您的服务器 运行 在 https 上吗？
请注意 navigator.getUserMedia 已弃用，您应该使用基于承诺的 navigator.mediaDevices.getUserMedia - Safari 仅支持后者。
我尝试与一些人建立一个视频组会议流，我使用 WebRTC（来自这里：https://github.com/anoek/webrtc-group-chat-example），作为服务器，我使用 VPS 安装了节点 js 的 Centos 7，当我尝试启动我的 webrtc 服务器，在浏览器中，我在使用端口
访问 ip 时看到此错误这个错误：
(index):260 Uncaught TypeError: navigator.getUserMedia is not a function
at setup_local_media ((index):260)
at Socket.<anonymous> ((index):50)
at Socket.Emitter.emit (socket.io-1.4.5.js:3)
at Socket.emit (socket.io-1.4.5.js:3)
at Socket.onconnect (socket.io-1.4.5.js:3)
at Socket.onpacket (socket.io-1.4.5.js:3)
at Manager.<anonymous> (socket.io-1.4.5.js:3)
at Manager.Emitter.emit (socket.io-1.4.5.js:3)
at Manager.ondecoded (socket.io-1.4.5.js:2)
at Decoder.<anonymous> (socket.io-1.4.5.js:3)
JS代码：
/*************/
/*** SETUP ***/
/*************/
var express = require('express');
var bodyParser = require('body-parser')
var cors = require('cors')
var main = express()
main.use(cors())
var server = main.listen(8000);
var io = require('socket.io').listen(server);
main.get('/', function(req, res){ res.sendFile(__dirname + '/client.html'); });
/*************************/
/*** INTERESTING STUFF ***/
/*************************/
var channels = {};
var sockets = {};
/**
* Users will connect to the signaling server, after which they'll issue a "join"
* to join a particular channel. The signaling server keeps track of all sockets
* who are in a channel, and on join will send out 'addPeer' events to each pair
* of users in a channel. When clients receive the 'addPeer' even they'll begin
* setting up an RTCPeerConnection with one another. During this process they'll
* need to relay ICECandidate information to one another, as well as SessionDescription
* information. After all of that happens, they'll finally be able to complete
* the peer connection and will be streaming audio/video between eachother.
*/
io.sockets.on('connection', function (socket) {
socket.channels = {};
sockets[socket.id] = socket;
console.log("["+ socket.id + "] connection accepted");
socket.on('disconnect', function () {
for (var channel in socket.channels) {
part(channel);
}
console.log("["+ socket.id + "] disconnected");
delete sockets[socket.id];
});
socket.on('join', function (config) {
console.log("["+ socket.id + "] join ", config);
var channel = config.channel;
var userdata = config.userdata;
if (channel in socket.channels) {
console.log("["+ socket.id + "] ERROR: already joined ", channel);
return;
}
if (!(channel in channels)) {
channels[channel] = {};
}
for (id in channels[channel]) {
channels[channel][id].emit('addPeer', {'peer_id': socket.id, 'should_create_offer': false});
socket.emit('addPeer', {'peer_id': id, 'should_create_offer': true});
}
channels[channel][socket.id] = socket;
socket.channels[channel] = channel;
});
function part(channel) {
console.log("["+ socket.id + "] part ");
if (!(channel in socket.channels)) {
console.log("["+ socket.id + "] ERROR: not in ", channel);
return;
}
delete socket.channels[channel];
delete channels[channel][socket.id];
for (id in channels[channel]) {
channels[channel][id].emit('removePeer', {'peer_id': socket.id});
socket.emit('removePeer', {'peer_id': id});
}
}
socket.on('part', part);
socket.on('relayICECandidate', function(config) {
var peer_id = config.peer_id;
var ice_candidate = config.ice_candidate;
console.log("["+ socket.id + "] relaying ICE candidate to [" + peer_id + "] ", ice_candidate);
if (peer_id in sockets) {
sockets[peer_id].emit('iceCandidate', {'peer_id': socket.id, 'ice_candidate': ice_candidate});
}
});
socket.on('relaySessionDescription', function(config) {
var peer_id = config.peer_id;
var session_description = config.session_description;
console.log("["+ socket.id + "] relaying session description to [" + peer_id + "] ", session_description);
if (peer_id in sockets) {
sockets[peer_id].emit('sessionDescription', {'peer_id': socket.id, 'session_description': session_description});
}
});
});
Html代码：
<!doctype html>
<html>
<head>
<script src="http://ajax.googleapis.com/ajax/libs/jquery/1.10.2/jquery.min.js"></script>
<!-- This adapter.js file de-prefixes the webkit* and moz* prefixed RTC
methods. When RTC becomes a more solid standard, this adapter should no
longer be necessary. -->
<!-- <script src="https://webrtc.googlecode.com/svn/trunk/samples/js/base/adapter.js"></script> -->
<style>
html, body {
background-color: #333;
}
video {
width: 320px;
height: 240px;
border: 1px solid black;
}
</style>
<script src="https://cdn.socket.io/socket.io-1.4.5.js"></script>
<script>
/** CONFIG **/
var SIGNALING_SERVER = "http://127.0.0.1:8000";
var USE_AUDIO = true;
var USE_VIDEO = true;
var DEFAULT_CHANNEL = 'some-global-channel-name';
var MUTE_AUDIO_BY_DEFAULT = false;
/** You should probably use a different stun server doing commercial stuff **/
/** Also see: https://gist.github.com/zziuni/3741933 **/
var ICE_SERVERS = [
{url:"stun:stun.l.google.com:19302"}
];
</script>
<script>
var signaling_socket = null; /* our socket.io connection to our webserver */
var local_media_stream = null; /* our own microphone / webcam */
var peers = {}; /* keep track of our peer connections, indexed by peer_id (aka socket.io id) */
var peer_media_elements = {}; /* keep track of our <video>/<audio> tags, indexed by peer_id */
function init() {
console.log("Connecting to signaling server");
signaling_socket = io(SIGNALING_SERVER);
signaling_socket = io();
signaling_socket.on('connect', function() {
console.log("Connected to signaling server");
setup_local_media(function() {
/* once the user has given us access to their
* microphone/camcorder, join the channel and start peering up */
join_chat_channel(DEFAULT_CHANNEL, {'whatever-you-want-here': 'stuff'});
});
});
signaling_socket.on('disconnect', function() {
console.log("Disconnected from signaling server");
/* Tear down all of our peer connections and remove all the
* media divs when we disconnect */
for (peer_id in peer_media_elements) {
peer_media_elements[peer_id].remove();
}
for (peer_id in peers) {
peers[peer_id].close();
}
peers = {};
peer_media_elements = {};
});
function join_chat_channel(channel, userdata) {
signaling_socket.emit('join', {"channel": channel, "userdata": userdata});
}
function part_chat_channel(channel) {
signaling_socket.emit('part', channel);
}
/**
* When we join a group, our signaling server will send out 'addPeer' events to each pair
* of users in the group (creating a fully-connected graph of users, ie if there are 6 people
* in the channel you will connect directly to the other 5, so there will be a total of 15
* connections in the network).
*/
signaling_socket.on('addPeer', function(config) {
console.log('Signaling server said to add peer:', config);
var peer_id = config.peer_id;
if (peer_id in peers) {
/* This could happen if the user joins multiple channels where the other peer is also in. */
console.log("Already connected to peer ", peer_id);
return;
}
var peer_connection = new RTCPeerConnection(
{"iceServers": ICE_SERVERS},
{"optional": [{"DtlsSrtpKeyAgreement": true}]} /* this will no longer be needed by chrome
* eventually (supposedly), but is necessary
* for now to get firefox to talk to chrome */
);
peers[peer_id] = peer_connection;
peer_connection.onicecandidate = function(event) {
if (event.candidate) {
signaling_socket.emit('relayICECandidate', {
'peer_id': peer_id,
'ice_candidate': {
'sdpMLineIndex': event.candidate.sdpMLineIndex,
'candidate': event.candidate.candidate
}
});
}
}
peer_connection.onaddstream = function(event) {
console.log("onAddStream", event);
var remote_media = USE_VIDEO ? $("<video>") : $("<audio>");
remote_media.attr("autoplay", "autoplay");
if (MUTE_AUDIO_BY_DEFAULT) {
remote_media.attr("muted", "true");
}
remote_media.attr("controls", "");
peer_media_elements[peer_id] = remote_media;
$('body').append(remote_media);
attachMediaStream(remote_media[0], event.stream);
}
/* Add our local stream */
peer_connection.addStream(local_media_stream);
/* Only one side of the peer connection should create the
* offer, the signaling server picks one to be the offerer.
* The other user will get a 'sessionDescription' event and will
* create an offer, then send back an answer 'sessionDescription' to us
*/
if (config.should_create_offer) {
console.log("Creating RTC offer to ", peer_id);
peer_connection.createOffer(
function (local_description) {
console.log("Local offer description is: ", local_description);
peer_connection.setLocalDescription(local_description,
function() {
signaling_socket.emit('relaySessionDescription',
{'peer_id': peer_id, 'session_description': local_description});
console.log("Offer setLocalDescription succeeded");
},
function() { Alert("Offer setLocalDescription failed!"); }
);
},
function (error) {
console.log("Error sending offer: ", error);
});
}
});
/**
* Peers exchange session descriptions which contains information
* about their audio / video settings and that sort of stuff. First
* the 'offerer' sends a description to the 'answerer' (with type
* "offer"), then the answerer sends one back (with type "answer").
*/
signaling_socket.on('sessionDescription', function(config) {
console.log('Remote description received: ', config);
var peer_id = config.peer_id;
var peer = peers[peer_id];
var remote_description = config.session_description;
console.log(config.session_description);
var desc = new RTCSessionDescription(remote_description);
var stuff = peer.setRemoteDescription(desc,
function() {
console.log("setRemoteDescription succeeded");
if (remote_description.type == "offer") {
console.log("Creating answer");
peer.createAnswer(
function(local_description) {
console.log("Answer description is: ", local_description);
peer.setLocalDescription(local_description,
function() {
signaling_socket.emit('relaySessionDescription',
{'peer_id': peer_id, 'session_description': local_description});
console.log("Answer setLocalDescription succeeded");
},
function() { Alert("Answer setLocalDescription failed!"); }
);
},
function(error) {
console.log("Error creating answer: ", error);
console.log(peer);
});
}
},
function(error) {
console.log("setRemoteDescription error: ", error);
}
);
console.log("Description Object: ", desc);
});
/**
* The offerer will send a number of ICE Candidate blobs to the answerer so they
* can begin trying to find the best path to one another on the net.
*/
signaling_socket.on('iceCandidate', function(config) {
var peer = peers[config.peer_id];
var ice_candidate = config.ice_candidate;
peer.addIceCandidate(new RTCIceCandidate(ice_candidate));
});
/**
* When a user leaves a channel (or is disconnected from the
* signaling server) everyone will recieve a 'removePeer' message
* telling them to trash the media channels they have open for those
* that peer. If it was this client that left a channel, they'll also
* receive the removePeers. If this client was disconnected, they
* wont receive removePeers, but rather the
* signaling_socket.on('disconnect') code will kick in and tear down
* all the peer sessions.
*/
signaling_socket.on('removePeer', function(config) {
console.log('Signaling server said to remove peer:', config);
var peer_id = config.peer_id;
if (peer_id in peer_media_elements) {
peer_media_elements[peer_id].remove();
}
if (peer_id in peers) {
peers[peer_id].close();
}
delete peers[peer_id];
delete peer_media_elements[config.peer_id];
});
}
/***********************/
/** Local media stuff **/
/***********************/
function setup_local_media(callback, errorback) {
if (local_media_stream != null) { /* ie, if we've already been initialized */
if (callback) callback();
return;
}
/* Ask user for permission to use the computers microphone and/or camera,
* attach it to an <audio> or <video> tag if they give us access. */
console.log("Requesting access to local audio / video inputs");
navigator.getUserMedia = ( navigator.getUserMedia ||
navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia ||
navigator.msGetUserMedia);
attachMediaStream = function(element, stream) {
console.log('DEPRECATED, attachMediaStream will soon be removed.');
element.srcObject = stream;
};
navigator.getUserMedia({"audio":USE_AUDIO, "video":USE_VIDEO},
function(stream) { /* user accepted access to a/v */
console.log("Access granted to audio/video");
local_media_stream = stream;
var local_media = USE_VIDEO ? $("<video>") : $("<audio>");
local_media.attr("autoplay", "autoplay");
local_media.attr("muted", "true"); /* always mute ourselves by default */
local_media.attr("controls", "");
$('body').append(local_media);
attachMediaStream(local_media[0], stream);
if (callback) callback();
},
function() { /* user denied access to a/v */
console.log("Access denied for audio/video");
alert("You chose not to provide access to the camera/microphone, demo will not work.");
if (errorback) errorback();
});
}
</script>
</head>
<body onload='init()'>
<!--
the <video> and <audio> tags are all added and removed dynamically
in 'onAddStream', 'setup_local_media', and 'removePeer'/'disconnect'
-->
</body>
</html>
navigator.getUserMedia 仅在安全上下文（即 https 或本地主机）中可用 - 您的服务器 运行 在 https 上吗？
请注意 navigator.getUserMedia 已弃用，您应该使用基于承诺的 navigator.mediaDevices.getUserMedia - Safari 仅支持后者。