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SquashSRS4: Add one to one RTC demo.

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winlin 2021-05-02 21:46:41 +08:00
parent 74bb47c13f
commit 206d95879f
101 changed files with 18902 additions and 9 deletions
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<html>
<head>
<title>SRS</title>
<meta charset="utf-8">
</head>
<body>
<h3><a href="https://github.com/ossrs/signaling">Signaling</a> works!</h3>
<p>
Run demo for <a href="one2one.html">WebRTC: One to One over SFU(SRS)</a><br/>
点击进入<a href="one2one.html">SRS一对一通话演示</a>
</p>
</body>

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/**
* The MIT License (MIT)
*
* Copyright (c) 2013-2021 Winlin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
* the Software, and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
'use strict';
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher.
function SrsRtcPublisherAsync() {
var self = {};
// @see https://github.com/rtcdn/rtcdn-draft
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// or autostart the publish:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
// webrtc://r.ossrs.net:11985/myapp/livestream
// or change the stream from livestream to mystream:
// webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream
// or set the candidate(ip) of answer:
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
// or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP:
// webrtc://r.ossrs.net/live/livestream?encrypt=false
// or any other information, will pass-by in the query:
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
// webrtc://r.ossrs.net/live/livestream?token=xxx
self.publish = async function (url) {
var conf = self.__internal.prepareUrl(url);
self.pc.addTransceiver("audio", {direction: "sendonly"});
self.pc.addTransceiver("video", {direction: "sendonly"});
var stream = await navigator.mediaDevices.getUserMedia(
{audio: true, video: {height: {max: 320}}}
);
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
stream.getTracks().forEach(function (track) {
self.pc.addTrack(track);
});
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
var session = await new Promise(function (resolve, reject) {
// @see https://github.com/rtcdn/rtcdn-draft
var data = {
api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
$.ajax({
type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
contentType: 'application/json', dataType: 'json'
}).done(function (data) {
console.log("Got answer: ", data);
if (data.code) {
reject(data);
return;
}
resolve(data);
}).fail(function (reason) {
reject(reason);
});
});
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
);
session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
// Notify about local stream when success.
self.onaddstream && self.onaddstream({stream: stream});
return session;
};
// Close the publisher.
self.close = function () {
self.pc && self.pc.close();
self.pc = null;
};
// The callback when got local stream.
self.onaddstream = function (event) {
};
// Internal APIs.
self.__internal = {
defaultPath: '/rtc/v1/publish/',
prepareUrl: function (webrtcUrl) {
var urlObject = self.__internal.parse(webrtcUrl);
// If user specifies the schema, use it as API schema.
var schema = urlObject.user_query.schema;
schema = schema ? schema + ':' : window.location.protocol;
var port = urlObject.port || 1985;
if (schema === 'https:') {
port = urlObject.port || 443;
}
// @see https://github.com/rtcdn/rtcdn-draft
var api = urlObject.user_query.play || self.__internal.defaultPath;
if (api.lastIndexOf('/') !== api.length - 1) {
api += '/';
}
apiUrl = schema + '//' + urlObject.server + ':' + port + api;
for (var key in urlObject.user_query) {
if (key !== 'api' && key !== 'play') {
apiUrl += '&' + key + '=' + urlObject.user_query[key];
}
}
// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
var apiUrl = apiUrl.replace(api + '&', api + '?');
var streamUrl = urlObject.url;
return {
apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
tid: new Date().getTime().toString(16)
};
},
parse: function (url) {
// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
var a = document.createElement("a");
a.href = url.replace("rtmp://", "http://")
.replace("webrtc://", "http://")
.replace("rtc://", "http://");
var vhost = a.hostname;
var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
// parse the vhost in the params of app, that srs supports.
app = app.replace("...vhost...", "?vhost=");
if (app.indexOf("?") >= 0) {
var params = app.substr(app.indexOf("?"));
app = app.substr(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.substr(0, vhost.indexOf("&"));
}
}
}
// when vhost equals to server, and server is ip,
// the vhost is __defaultVhost__
if (a.hostname === vhost) {
var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
if (re.test(a.hostname)) {
vhost = "__defaultVhost__";
}
}
// parse the schema
var schema = "rtmp";
if (url.indexOf("://") > 0) {
schema = url.substr(0, url.indexOf("://"));
}
var port = a.port;
if (!port) {
if (schema === 'http') {
port = 80;
} else if (schema === 'https') {
port = 443;
} else if (schema === 'rtmp') {
port = 1935;
}
}
var ret = {
url: url,
schema: schema,
server: a.hostname, port: port,
vhost: vhost, app: app, stream: stream
};
self.__internal.fill_query(a.search, ret);
// For webrtc API, we use 443 if page is https, or schema specified it.
if (!ret.port) {
if (schema === 'webrtc' || schema === 'rtc') {
if (ret.user_query.schema === 'https') {
ret.port = 443;
} else if (window.location.href.indexOf('https://') === 0) {
ret.port = 443;
} else {
// For WebRTC, SRS use 1985 as default API port.
ret.port = 1985;
}
}
}
return ret;
},
fill_query: function (query_string, obj) {
// pure user query object.
obj.user_query = {};
if (query_string.length === 0) {
return;
}
// split again for angularjs.
if (query_string.indexOf("?") >= 0) {
query_string = query_string.split("?")[1];
}
var queries = query_string.split("&");
for (var i = 0; i < queries.length; i++) {
var elem = queries[i];
var query = elem.split("=");
obj[query[0]] = query[1];
obj.user_query[query[0]] = query[1];
}
// alias domain for vhost.
if (obj.domain) {
obj.vhost = obj.domain;
}
}
};
self.pc = new RTCPeerConnection(null);
return self;
}
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-await-promise based SRS RTC Player.
function SrsRtcPlayerAsync() {
var self = {};
// @see https://github.com/rtcdn/rtcdn-draft
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// or autostart the play:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
// webrtc://r.ossrs.net:11985/myapp/livestream
// or change the stream from livestream to mystream:
// webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream
// or set the candidate(ip) of answer:
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
// or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP:
// webrtc://r.ossrs.net/live/livestream?encrypt=false
// or any other information, will pass-by in the query:
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
// webrtc://r.ossrs.net/live/livestream?token=xxx
self.play = async function(url) {
var conf = self.__internal.prepareUrl(url);
self.pc.addTransceiver("audio", {direction: "recvonly"});
self.pc.addTransceiver("video", {direction: "recvonly"});
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
var session = await new Promise(function(resolve, reject) {
// @see https://github.com/rtcdn/rtcdn-draft
var data = {
api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
$.ajax({
type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
contentType:'application/json', dataType: 'json'
}).done(function(data) {
console.log("Got answer: ", data);
if (data.code) {
reject(data); return;
}
resolve(data);
}).fail(function(reason){
reject(reason);
});
});
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
);
return session;
};
// Close the player.
self.close = function() {
self.pc && self.pc.close();
self.pc = null;
};
// The callback when got remote stream.
self.onaddstream = function (event) {};
// Internal APIs.
self.__internal = {
defaultPath: '/rtc/v1/play/',
prepareUrl: function (webrtcUrl) {
var urlObject = self.__internal.parse(webrtcUrl);
// If user specifies the schema, use it as API schema.
var schema = urlObject.user_query.schema;
schema = schema ? schema + ':' : window.location.protocol;
var port = urlObject.port || 1985;
if (schema === 'https:') {
port = urlObject.port || 443;
}
// @see https://github.com/rtcdn/rtcdn-draft
var api = urlObject.user_query.play || self.__internal.defaultPath;
if (api.lastIndexOf('/') !== api.length - 1) {
api += '/';
}
apiUrl = schema + '//' + urlObject.server + ':' + port + api;
for (var key in urlObject.user_query) {
if (key !== 'api' && key !== 'play') {
apiUrl += '&' + key + '=' + urlObject.user_query[key];
}
}
// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
var apiUrl = apiUrl.replace(api + '&', api + '?');
var streamUrl = urlObject.url;
return {
apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
tid: new Date().getTime().toString(16)
};
},
parse: function (url) {
// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
var a = document.createElement("a");
a.href = url.replace("rtmp://", "http://")
.replace("webrtc://", "http://")
.replace("rtc://", "http://");
var vhost = a.hostname;
var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
// parse the vhost in the params of app, that srs supports.
app = app.replace("...vhost...", "?vhost=");
if (app.indexOf("?") >= 0) {
var params = app.substr(app.indexOf("?"));
app = app.substr(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.substr(0, vhost.indexOf("&"));
}
}
}
// when vhost equals to server, and server is ip,
// the vhost is __defaultVhost__
if (a.hostname === vhost) {
var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
if (re.test(a.hostname)) {
vhost = "__defaultVhost__";
}
}
// parse the schema
var schema = "rtmp";
if (url.indexOf("://") > 0) {
schema = url.substr(0, url.indexOf("://"));
}
var port = a.port;
if (!port) {
if (schema === 'http') {
port = 80;
} else if (schema === 'https') {
port = 443;
} else if (schema === 'rtmp') {
port = 1935;
}
}
var ret = {
url: url,
schema: schema,
server: a.hostname, port: port,
vhost: vhost, app: app, stream: stream
};
self.__internal.fill_query(a.search, ret);
// For webrtc API, we use 443 if page is https, or schema specified it.
if (!ret.port) {
if (schema === 'webrtc' || schema === 'rtc') {
if (ret.user_query.schema === 'https') {
ret.port = 443;
} else if (window.location.href.indexOf('https://') === 0) {
ret.port = 443;
} else {
// For WebRTC, SRS use 1985 as default API port.
ret.port = 1985;
}
}
}
return ret;
},
fill_query: function (query_string, obj) {
// pure user query object.
obj.user_query = {};
if (query_string.length === 0) {
return;
}
// split again for angularjs.
if (query_string.indexOf("?") >= 0) {
query_string = query_string.split("?")[1];
}
var queries = query_string.split("&");
for (var i = 0; i < queries.length; i++) {
var elem = queries[i];
var query = elem.split("=");
obj[query[0]] = query[1];
obj.user_query[query[0]] = query[1];
}
// alias domain for vhost.
if (obj.domain) {
obj.vhost = obj.domain;
}
}
};
self.pc = new RTCPeerConnection(null);
self.pc.onaddstream = function (event) {
if (self.onaddstream) {
self.onaddstream(event);
}
};
return self;
}
// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
function SrsRtcFormatSenders(senders, kind) {
var codecs = [];
senders.forEach(function (sender) {
sender.getParameters().codecs.forEach(function(c) {
if (kind && sender.track.kind !== kind) {
return;
}
if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
return;
}
var s = '';
s += c.mimeType.replace('audio/', '').replace('video/', '');
s += ', ' + c.clockRate + 'HZ';
if (sender.track.kind === "audio") {
s += ', channels: ' + c.channels;
}
s += ', pt: ' + c.payloadType;
codecs.push(s);
});
});
return codecs.join(", ");
}

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trunk/3rdparty/signaling/www/demos/js/srs.sig.js vendored Normal file
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/**
* The MIT License (MIT)
*
* Copyright (c) 2013-2021 Winlin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
* the Software, and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
'use strict';
// Async-await-promise based SRS RTC Signaling.
function SrsRtcSignalingAsync() {
var self = {};
// The schema is ws or wss, host is ip or ip:port, display is nickname
// of user to join the room.
self.connect = async function (schema, host, room, display) {
var url = schema + '://' + host + '/sig/v1/rtc';
self.ws = new WebSocket(url + '?room=' + room + '&display=' + display);
self.ws.onmessage = function(event) {
var r = JSON.parse(event.data);
var promise = self._internals.msgs[r.tid];
if (promise) {
promise.resolve(r.msg);
delete self._internals.msgs[r.tid];
} else {
self.onmessage(r.msg);
}
};
return new Promise(function (resolve, reject) {
self.ws.onopen = function (event) {
resolve(event);
};
self.ws.onerror = function (event) {
reject(event);
};
});
};
// The message is a json object.
self.send = async function (message) {
return new Promise(function (resolve, reject) {
var r = {tid: new Date().getTime().toString(16), msg: message};
self._internals.msgs[r.tid] = {resolve: resolve, reject: reject};
self.ws.send(JSON.stringify(r));
});
};
self.close = function () {
self.ws && self.ws.close();
self.ws = null;
for (const tid in self._internals.msgs) {
var promise = self._internals.msgs[tid];
promise.reject('close');
}
};
// The callback when got messages from signaling server.
self.onmessage = function (msg) {
};
self._internals = {
// Key is tid, value is object {resolve, reject, response}.
msgs: {}
};
return self;
}
// Parse params in query string.
function SrsRtcSignalingParse(location) {
let query = location.href.split('?')[1];
query = query? '?' + query : null;
let wsSchema = location.href.split('wss=')[1];
wsSchema = wsSchema? wsSchema.split('&')[0] : (location.protocol === 'http:'? 'ws' : 'wss');
let wsHost = location.href.split('wsh=')[1];
wsHost = wsHost? wsHost.split('&')[0] : location.hostname;
let wsPort = location.href.split('wsp=')[1];
wsPort = wsPort? wsPort.split('&')[0] : location.host.split(':')[1];
wsHost = wsPort? wsHost.split(':')[0] + ':' + wsPort : wsHost;
let host = location.href.split('host=')[1];
host = host? host.split('&')[0] : location.hostname;
let room = location.href.split('room=')[1];
let display = location.href.split('display=')[1];
display = display? display.split('&')[0] : new Date().getTime().toString(16).substr(3);
let autostart = location.href.split('autostart=')[1];
autostart = autostart && autostart.split('&')[0] === 'true';
return {
query: query, wsSchema: wsSchema, wsHost: wsHost, host: host,
room: room, display: display, autostart: autostart,
};
}

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<!DOCTYPE html>
<html>
<head>
<title>SRS</title>
<meta charset="utf-8">
<style>
body{
padding-top: 55px;
}
</style>
<link rel="stylesheet" type="text/css" href="css/bootstrap.min.css"/>
<script type="text/javascript" src="js/jquery-1.10.2.min.js"></script>
<script type="text/javascript" src="js/adapter-7.4.0.min.js"></script>
<script type="text/javascript" src="js/srs.sdk.js"></script>
<script type="text/javascript" src="js/srs.sig.js"></script>
</head>
<body>
<img src='https://ossrs.net/gif/v1/sls.gif?site=ossrs.net&path=/player/rtcpublisher'/>
<div class="navbar navbar-fixed-top">
<div class="navbar-inner">
<div class="container">
<a class="brand" href="https://github.com/ossrs/srs">SRS</a>
<div class="nav-collapse collapse">
<ul class="nav">
<li class="active"><a href="#">一对一通话</a></li>
<li>
<a href="https://github.com/ossrs/signaling">
<img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/signaling?style=social">
</a>
</li>
</ul>
</div>
</div>
</div>
</div>
<div class="container">
<div class="form-inline">
SRS:
<input type="text" id="txt_host" class="input-medium" value="">
Room:
<input type="text" id="txt_room" class="input-small" value="live">
Display:
<input type="text" id="txt_display" class="input-small" value="">
<button class="btn btn-primary" id="btn_start">开始通话</button>
</div>
<div class="row">
<div class="span5">
<label></label>
<video id="rtc_media_publisher" width="320" autoplay muted controls></video>
<label></label>
<span id='self'></span>
</div>
<div class="span6">
<label></label>
<video id="rtc_media_player" width="320" autoplay muted controls></video>
<label></label>
<span id='peer'></span>
</div>
</div>
</div>
<script type="text/javascript">
var sig = null;
var publisher = null;
var player = null;
$(function(){
console.log('?wss=x to specify the websocket schema, ws or wss');
console.log('?wsh=x to specify the websocket server ip');
console.log('?wsp=x to specify the websocket server port');
console.log('?host=x to specify the SRS server');
console.log('?room=x to specify the room to join');
console.log('?display=x to specify your nick name');
var startDemo = async function () {
var host = $('#txt_host').val();
var room = $('#txt_room').val();
var display = $('#txt_display').val();
// Connect to signaling first.
if (sig) {
sig.close();
}
sig = new SrsRtcSignalingAsync();
sig.onmessage = function (msg) {
console.log('Notify: ', msg);
msg.participants.forEach(function (participant) {
if (participant.display === display || !participant.publishing) return;
startPlay(host, room, participant.display);
});
};
await sig.connect(conf.wsSchema, conf.wsHost, room, display);
let r0 = await sig.send({action:'join', room:room, display:display});
console.log('Signaling: join ok', r0);
// For one to one demo, alert and ignore when room is full.
if (r0.participants.length > 2) {
alert('Room is full, already ' + (r0.participants.length - 1) + ' participants');
sig.close();
return;
}
// Start publish media if signaling is ok.
await startPublish(host, room, display);
let r1 = await sig.send({action:'publish', room:room, display:display});
console.log('Signaling: publish ok', r1);
// Play the stream already in room.
r0.participants.forEach(function(participant) {
if (participant.display === display || !participant.publishing) return;
startPlay(host, room, participant.display);
});
};
var startPublish = function (host, room, display) {
var url = 'webrtc://' + host + '/' + room + '/' + display + conf.query;
$('#rtc_media_publisher').show();
if (publisher) {
publisher.close();
}
publisher = new SrsRtcPublisherAsync();
publisher.onaddstream = function (event) {
console.log('Start publish, event: ', event);
$('#rtc_media_publisher').prop('srcObject', event.stream);
};
return publisher.publish(url).then(function(session){
$('#self').text('Self: ' + display);
}).catch(function (reason) {
publisher.close();
$('#rtc_media_publisher').hide();
console.error(reason);
});
};
var startPlay = function (host, room, display) {
var url = 'webrtc://' + host + '/' + room + '/' + display + conf.query;
$('#rtc_media_player').show();
if (player) {
player.close();
}
player = new SrsRtcPlayerAsync();
player.onaddstream = function (event) {
console.log('Start play, event: ', event);
$('#rtc_media_player').prop('srcObject', event.stream);
};
player.play(url).then(function(session){
$('#peer').text('Peer: ' + display);
$('#rtc_media_player').prop('muted', false);
}).catch(function (reason) {
player.close();
$('#rtc_media_player').hide();
console.error(reason);
});
};
$('#rtc_media_publisher').hide();
$('#rtc_media_player').hide();
$("#btn_start").click(startDemo);
// Pass-by to SRS url.
let conf = SrsRtcSignalingParse(window.location);
$('#txt_host').val(conf.host);
conf.room && $('#txt_room').val(conf.room);
$('#txt_display').val(conf.display);
if (conf.autostart) {
startDemo();
}
});
</script>
</body>
</html>