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325 lines
13 KiB
HTML
325 lines
13 KiB
HTML
<!DOCTYPE html>
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<html>
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<head>
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<title>SRS</title>
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<meta charset="utf-8">
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<style>
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body{
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padding-top: 55px;
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}
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</style>
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<link rel="stylesheet" type="text/css" href="css/bootstrap.min.css"/>
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<script type="text/javascript" src="js/jquery-1.10.2.min.js"></script>
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<script type="text/javascript" src="js/adapter-7.4.0.min.js"></script>
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<script type="text/javascript" src="js/winlin.utility.js"></script>
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<script type="text/javascript" src="js/srs.page.js"></script>
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</head>
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<body>
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<img src='https://ossrs.net/gif/v1/sls.gif?site=ossrs.net&path=/player/rtcplayer'/>
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<div class="navbar navbar-fixed-top">
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<div class="navbar-inner">
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<div class="container">
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<a id="srs_index" class="brand" href="https://github.com/ossrs/srs">SRS</a>
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<div class="nav-collapse collapse">
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<ul class="nav">
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<li><a id="nav_srs_player" href="srs_player.html">SRS播放器</a></li>
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<li class="active"><a id="nav_rtc_player" href="rtc_player.html">RTC播放器</a></li>
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<li><a id="nav_rtc_publisher" href="rtc_publisher.html">RTC推流</a></li>
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<li><a href="http://ossrs.net/srs.release/releases/app.html">iOS/Andriod</a></li>
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<!--<li><a id="nav_srs_publisher" href="srs_publisher.html">SRS编码器</a></li>-->
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<!--<li><a id="nav_srs_chat" href="srs_chat.html">SRS会议</a></li>-->
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<!--<li><a id="nav_srs_bwt" href="srs_bwt.html">SRS测网速</a></li>-->
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<!--<li><a id="nav_vlc" href="vlc.html">VLC播放器</a></li>-->
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<li><a id="nav_gb28181" href="srs_gb28181.html">GB28181</a></li>
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<li>
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<a href="https://github.com/ossrs/srs">
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<img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/srs?style=social">
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</a>
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</li>
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</ul>
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</div>
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</div>
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</div>
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</div>
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<div class="container">
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<div class="form-inline">
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URL:
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<input type="text" id="txt_url" class="input-xxlarge" value="">
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<button class="btn btn-primary" id="btn_play">播放视频</button>
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</div>
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<label></label>
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<video id="rtc_media_player" controls autoplay></video>
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<label></label>
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SessionID: <span id='sessionid'></span>
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<label></label>
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Simulator: <a href='#' id='simulator-drop'>Drop</a>
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<footer>
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<p></p>
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<p><a href="https://github.com/ossrs/srs">SRS Team © 2020</a></p>
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</footer>
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</div>
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<script type="text/javascript">
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$(function(){
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// Async-await-promise based SRS RTC Player.
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function SrsRtcPlayerAsync() {
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var self = {};
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// @see https://github.com/rtcdn/rtcdn-draft
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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// or specifies the API port:
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// webrtc://r.ossrs.net:11985/live/livestream
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// or autostart the play:
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// webrtc://r.ossrs.net/live/livestream?autostart=true
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// or change the app from live to myapp:
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// webrtc://r.ossrs.net:11985/myapp/livestream
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// or change the stream from livestream to mystream:
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(ip) of answer:
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// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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// webrtc://r.ossrs.net/live/livestream?encrypt=false
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// or any other information, will pass-by in the query:
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// webrtc://r.ossrs.net/live/livestream?vhost=xxx
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// webrtc://r.ossrs.net/live/livestream?token=xxx
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self.play = async function(url) {
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var conf = self.__internal.prepareUrl(url);
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self.pc.addTransceiver("audio", {direction: "recvonly"});
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self.pc.addTransceiver("video", {direction: "recvonly"});
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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var session = await new Promise(function(resolve, reject) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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$.ajax({
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type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
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contentType:'application/json', dataType: 'json'
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}).done(function(data) {
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console.log("Got answer: ", data);
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if (data.code) {
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reject(data); return;
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}
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resolve(data);
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}).fail(function(reason){
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reject(reason);
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});
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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);
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return session;
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};
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// Close the publisher.
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self.close = function() {
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self.pc.close();
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};
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// The callback when got remote stream.
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self.onaddstream = function (event) {};
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// Internal APIs.
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self.__internal = {
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defaultPath: '/rtc/v1/play/',
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prepareUrl: function (webrtcUrl) {
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var urlObject = self.__internal.parse(webrtcUrl);
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// If user specifies the schema, use it as API schema.
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var schema = urlObject.user_query.schema;
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schema = schema ? schema + ':' : window.location.protocol;
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var port = urlObject.port || 1985;
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if (schema === 'https:') {
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port = urlObject.port || 443;
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}
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// @see https://github.com/rtcdn/rtcdn-draft
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var api = urlObject.user_query.play || self.__internal.defaultPath;
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if (api.lastIndexOf('/') !== api.length - 1) {
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api += '/';
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}
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apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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for (var key in urlObject.user_query) {
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if (key !== 'api' && key !== 'play') {
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apiUrl += '&' + key + '=' + urlObject.user_query[key];
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}
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}
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// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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var apiUrl = apiUrl.replace(api + '&', api + '?');
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var streamUrl = urlObject.url;
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return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
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},
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parse: function (url) {
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// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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var a = document.createElement("a");
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a.href = url.replace("rtmp://", "http://")
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.replace("webrtc://", "http://")
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.replace("rtc://", "http://");
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var vhost = a.hostname;
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var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
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var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
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// parse the vhost in the params of app, that srs supports.
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app = app.replace("...vhost...", "?vhost=");
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if (app.indexOf("?") >= 0) {
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var params = app.substr(app.indexOf("?"));
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app = app.substr(0, app.indexOf("?"));
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if (params.indexOf("vhost=") > 0) {
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vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
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if (vhost.indexOf("&") > 0) {
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vhost = vhost.substr(0, vhost.indexOf("&"));
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}
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}
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}
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// when vhost equals to server, and server is ip,
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// the vhost is __defaultVhost__
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if (a.hostname === vhost) {
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var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
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if (re.test(a.hostname)) {
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vhost = "__defaultVhost__";
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}
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}
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// parse the schema
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var schema = "rtmp";
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if (url.indexOf("://") > 0) {
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schema = url.substr(0, url.indexOf("://"));
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}
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var port = a.port;
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if (!port) {
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if (schema === 'http') {
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port = 80;
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} else if (schema === 'https') {
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port = 443;
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} else if (schema === 'rtmp') {
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port = 1935;
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}
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}
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var ret = {
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url: url,
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schema: schema,
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server: a.hostname, port: port,
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vhost: vhost, app: app, stream: stream
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};
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self.__internal.fill_query(a.search, ret);
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// For webrtc API, we use 443 if page is https, or schema specified it.
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if (!ret.port) {
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if (schema === 'webrtc' || schema === 'rtc') {
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if (ret.user_query.schema === 'https') {
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ret.port = 443;
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} else if (window.location.href.indexOf('https://') === 0) {
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ret.port = 443;
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} else {
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// For WebRTC, SRS use 1985 as default API port.
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ret.port = 1985;
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}
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}
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}
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return ret;
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},
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fill_query: function (query_string, obj) {
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// pure user query object.
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obj.user_query = {};
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if (query_string.length === 0) {
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return;
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}
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// split again for angularjs.
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if (query_string.indexOf("?") >= 0) {
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query_string = query_string.split("?")[1];
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}
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var queries = query_string.split("&");
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for (var i = 0; i < queries.length; i++) {
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var elem = queries[i];
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var query = elem.split("=");
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obj[query[0]] = query[1];
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obj.user_query[query[0]] = query[1];
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}
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// alias domain for vhost.
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if (obj.domain) {
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obj.vhost = obj.domain;
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}
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}
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};
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self.pc = new RTCPeerConnection(null);
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self.pc.onaddstream = function (event) {
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if (self.onaddstream) {
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self.onaddstream(event);
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}
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};
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return self;
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}
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var sdk = null; // Global handler to do cleanup when replaying.
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var startPlay = function() {
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$('#rtc_media_player').show();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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}
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sdk = new SrsRtcPlayerAsync();
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sdk.onaddstream = function (event) {
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console.log('Start play, event: ', event);
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$('#rtc_media_player').prop('srcObject', event.stream);
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};
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// For example:
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// webrtc://r.ossrs.net/live/livestream
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var url = $("#txt_url").val();
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sdk.play(url).then(function(session){
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$('#sessionid').html(session.sessionid);
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$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$('#rtc_media_player').hide();
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var query = parse_query_string();
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srs_init_rtc("#txt_url", query);
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$("#btn_play").click(startPlay);
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if (query.autostart === 'true') {
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// For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a
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$('#rtc_media_player').prop('muted', true);
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startPlay();
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}
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});
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</script>
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</body>
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</html>
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