third
/
srs
zrcadlo https://github.com/ossrs/srs


			
				
					
						
						
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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/winlin.utility.js"></script>
    <script type="text/javascript" src="js/srs.page.js"></script>
</head>
<body>
<img src='https://ossrs.net/gif/v1/sls.gif?site=ossrs.net&path=/player/rtcplayer'/>
<div class="navbar navbar-fixed-top">
    <div class="navbar-inner">
        <div class="container">
            <a id="srs_index" class="brand" href="https://github.com/ossrs/srs">SRS</a>
            <div class="nav-collapse collapse">
                <ul class="nav">
                    <li><a id="nav_srs_player" href="srs_player.html">SRS播放器</a></li>
                    <li class="active"><a id="nav_rtc_player" href="rtc_player.html">RTC播放器</a></li>
                    <li><a id="nav_rtc_publisher" href="rtc_publisher.html">RTC推流</a></li>
                    <li><a  href="http://ossrs.net/srs.release/releases/app.html">iOS/Andriod</a></li>
                    <!--<li><a id="nav_srs_publisher" href="srs_publisher.html">SRS编码器</a></li>-->
                    <!--<li><a id="nav_srs_chat" href="srs_chat.html">SRS会议</a></li>-->
                    <!--<li><a id="nav_srs_bwt" href="srs_bwt.html">SRS测网速</a></li>-->
                    <!--<li><a id="nav_vlc" href="vlc.html">VLC播放器</a></li>-->
                    <li><a id="nav_gb28181" href="srs_gb28181.html">GB28181</a></li>
                    <li>
                        <a href="https://github.com/ossrs/srs">
                            <img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/srs?style=social">
                        </a>
                    </li>
                </ul>
            </div>
        </div>
    </div>
</div>
<div class="container">
    <div class="form-inline">
        URL:
        <input type="text" id="txt_url" class="input-xxlarge" value="">
        <button class="btn btn-primary" id="btn_play">播放视频</button>
    </div>

    <label></label>
    <video id="rtc_media_player" controls autoplay></video>

    <label></label>
    SessionID: <span id='sessionid'></span>

    <label></label>
    Simulator: <a href='#' id='simulator-drop'>Drop</a>

    <footer>
        <p></p>
        <p><a href="https://github.com/ossrs/srs">SRS Team &copy; 2020</a></p>
    </footer>
</div>
<script type="text/javascript">
    $(function(){
        // 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, 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 publisher.
            self.close = function() {
                self.pc.close();
            };

            // 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};
                },
                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;
        }

        var sdk = null; // Global handler to do cleanup when replaying.
        var startPlay = function() {
            $('#rtc_media_player').show();

            // Close PC when user replay.
            if (sdk) {
                sdk.close();
            }

            sdk = new SrsRtcPlayerAsync();
            sdk.onaddstream = function (event) {
                console.log('Start play, event: ', event);
                $('#rtc_media_player').prop('srcObject', event.stream);
            };

            // For example:
            //      webrtc://r.ossrs.net/live/livestream
            var url = $("#txt_url").val();
            sdk.play(url).then(function(session){
                $('#sessionid').html(session.sessionid);
                $('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
            }).catch(function (reason) {
                sdk.close();
                $('#rtc_media_player').hide();
                console.error(reason);
            });
        };

        $('#rtc_media_player').hide();
        var query = parse_query_string();
        srs_init_rtc("#txt_url", query);

        $("#btn_play").click(function() {
            $('#rtc_media_player').prop('muted', false);
            startPlay();
        });

        if (query.autostart === 'true') {
            $('#rtc_media_player').prop('muted', true);
            console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
                'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');

            startPlay();
        }
    });
</script>
</body>
</html>