Plugin SrsPlayer for live streaming

* Embed live streaming into a post/page or anywhere on your WordPress site
* Supports HLS/HTTP-FLV/WebRTC over HTTP or HTTPS for live streaming
* Embed HTML5 videos which are compatible with all major browsers
* Automatically play a video when the page is rendered
* No setup required, simply install and start embedding videos
* Lightweight and compatible with the latest version of WordPress

Please see https://github.com/ossrs/Typecho-Plugin-SrsPlayer

SrsPlayer/Plugin.php

@@ -0,0 +1,174 @@
+<?php
+
+define( 'SRS_PLAYER_VERSION', '1.0.3' );
+
+/**
+ * SRS Player is a video streaming player, supports HLS/HTTP-FLV/WebRTC etc.
+ *
+ * @package SrsPlayer
+ * @author Winlin Yang
+ * @version 1.0.3
+ * @link https://github.com/ossrs/Typecho-Plugin-SrsPlayer
+ */
+class SrsPlayer_Plugin implements Typecho_Plugin_Interface
+{
+    public static function activate() {
+        Typecho_Plugin::factory('Widget_Archive')->header = array(__CLASS__, 'header');
+        Typecho_Plugin::factory('Widget_Archive')->footer = array(__CLASS__, 'footer');
+        Typecho_Plugin::factory('Widget_Abstract_Contents')->content = array('SrsPlayer_Plugin', 'parse');
+    }
+
+    public static function deactivate() {
+    }
+
+    public static function config(Typecho_Widget_Helper_Form $form) {
+    }
+
+    public static function personalConfig(Typecho_Widget_Helper_Form $form) {
+    }
+
+    public static function render() {
+    }
+
+    public static function header(){
+        echo "\n";
+        $cssUrl = Helper::options()->pluginUrl . '/SrsPlayer/public/css/srs-player-public.css?v=' . SRS_PLAYER_VERSION;
+        echo '<link rel="stylesheet" href="' . $cssUrl . '">' . "\n";
+
+        // We must preload the jQuery and SrsPlayer.
+        $urls = array(
+            Helper::options()->pluginUrl . '/SrsPlayer/public/js/jquery-1.10.2.min.js?v=' . SRS_PLAYER_VERSION,
+            Helper::options()->pluginUrl . '/SrsPlayer/public/js/srs.player.js?v=' . SRS_PLAYER_VERSION,
+        );
+        foreach ($urls as $url) {
+            echo "<script src='${url}'></script>\n";
+        }
+    }
+
+    public static function footer() {
+        // Lazy load after page is loaded.
+        $urls = array(
+            Helper::options()->pluginUrl . '/SrsPlayer/public/js/flv-1.5.0.min.js?v=' . SRS_PLAYER_VERSION,
+            Helper::options()->pluginUrl . '/SrsPlayer/public/js/hls-0.14.17.min.js?v=' . SRS_PLAYER_VERSION,
+            Helper::options()->pluginUrl . '/SrsPlayer/public/js/adapter-7.4.0.js?v=' . SRS_PLAYER_VERSION,
+            Helper::options()->pluginUrl . '/SrsPlayer/public/js/srs.sdk.js?v=' . SRS_PLAYER_VERSION,
+        );
+        foreach ($urls as $url) {
+            echo "<script src='${url}'></script>\n";
+        }
+    }
+
+    public static function parse($text, $widget, $lastResult) {
+        if (!($widget instanceof Widget_Archive)) {
+            return $text;
+        }
+
+        $matches = array();
+        preg_match_all('/\[(srs_player).*\]/', $text, $matches);
+        if (empty($matches) || empty($matches[0])) {
+            return $text;
+        }
+
+        $o = $text;
+        foreach ($matches[0] as $match) {
+            // The $match is the player instance, for example,
+            //      [srs_player url='https://r.ossrs.net/live/livestream.m3u8' muted width="720"]
+            // We parse $match to $obj as:
+            //      url: https://r.ossrs.net/live/livestream.m3u8
+            //      muted: muted
+            //      width: 720
+            $obj = SrsPlayer_Plugin::toObject($match);
+
+            // Build $obj to $replace, the new H5 element.
+            $replace = SrsPlayer_Plugin::buildReplacement($match, $obj);
+
+            // Replace the $match to the new H5 object.
+            $o = str_replace($match, $replace, $o);
+        }
+
+        return $o;
+    }
+
+    private static function toObject($match) {
+        $input = trim(rtrim(ltrim(ltrim(ltrim($match, "[")), "srs_player"), "]"));
+
+        $attrs = array(
+            'url' => '',
+            'controls' => 'controls',
+            'autoplay' => 'autoplay',
+            'muted' => 'muted',
+            'width' => '',
+        );
+
+        foreach (explode(" ", $input) as $e) {
+            $kv = explode("=", $e);
+            if (count($kv) == 0) {
+                continue;
+            }
+
+            $k = $kv[0];
+            if (count($kv) == 1) {
+                $attrs[$k] = $k;
+            } else {
+                $attrs[$k] = $kv[1];
+            }
+        }
+
+        foreach ($attrs as $k => $v) {
+            $attrs[$k] = trim(trim($v, '"'), "'");
+        }
+
+        return $attrs;
+    }
+
+    private static function buildReplacement($match, $q) {
+        if (empty($q)) {
+            return "Invalid ${match}";
+        }
+
+        if (empty($q['url'])) {
+            return "No URL of ${match}";
+        }
+
+        $url = $q['url'];
+        $id = 'srs-player-' . SrsPlayer_Plugin::random_str(32);
+
+        $controls = ' controls=' . $q['controls'];
+        if ($q['controls'] != 'true' && $q['controls'] != 'controls') $controls = '';
+
+        $autoplay = ' autoplay=' . $q['autoplay'];
+        if ($q['autoplay'] != 'true' && $q['autoplay'] != 'autoplay') $autoplay = '';
+
+        $muted = ' muted=' . $q['muted'];
+        if ($q['muted'] != 'true' && $q['muted'] != 'muted') $muted = '';
+
+        $width = ' width="' . $q['width'] . '"';
+        if (empty($q['width'])) $width = '';
+
+        $o = <<<EOT
+    <div class="srs-player-wrapper">
+        <video id="${id}" ${controls}${autoplay}${muted}${width}></video>
+        <script>(function($) { new SrsPlayer("#${id}", "${url}").play(); })(jQuery);</script>
+    </div>
+EOT;
+        return $o;
+    }
+
+    private static function random_str($length, $keyspace = NULL) {
+        if (empty($keyspace)) {
+            $keyspace = '0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ';
+        }
+
+        if ($length < 1) {
+            throw new RangeException("Invalid length ${length}");
+        }
+
+        $pieces = array();
+        $max = mb_strlen($keyspace, '8bit') - 1;
+        for ($i = 0; $i < $length; ++$i) {
+            $pieces []= $keyspace[rand(0, $max)];
+        }
+        return implode('', $pieces);
+    }
+}
+

SrsPlayer/public/css/srs-player-public.css

@@ -0,0 +1,6 @@
+/**
+ * Style for SRS Player
+ */
+.srs-player-wrapper {
+}
+

SrsPlayer/public/index.php

@@ -0,0 +1,3 @@
+<?php
+// Silence is golden
+

SrsPlayer/public/js/srs.player.js

@@ -0,0 +1,60 @@
+
+'use strict';
+
+(function($) {
+  window.SrsPlayer = function (dom, url) {
+    const self = {
+      dom: $(dom),
+      url,
+    };
+
+    self.play = function() {
+      if (document.readyState !== "complete") {
+        return setTimeout(self.play, 0);
+      }
+      self.__play();
+    }
+
+    self.__play = function () {
+      if (self.url.indexOf('.mp4') > 0) {
+        self.dom.prop('src', self.url);
+        return console.log(`Play by native for ${self.url}`);
+      }
+
+      if (self.url.indexOf('.flv') > 0) {
+        if (!flvjs.isSupported()) return console.error(`HTTP-FLV is not supported by browser`);
+
+        const player = flvjs.createPlayer({type: 'flv', url: self.url});
+        player.attachMediaElement(self.dom.get(0));
+        player.load();
+        player.play();
+        return console.log(`Play by flv.js for ${self.url}`);
+      }
+
+      if (self.url.indexOf('.m3u8') > 0) {
+        // See https://stackoverflow.com/a/12905122/17679565
+        if (document.createElement('video').canPlayType('application/vnd.apple.mpegURL')) {
+          self.dom.prop('src', self.url);
+          return console.log(`Play by native for ${self.url}`);
+        }
+
+        const player = new Hls();
+        player.loadSource(self.url);
+        player.attachMedia(self.dom.get(0));
+        return console.log(`Play by hls.js for ${self.url}`);
+      }
+
+      if (self.url.indexOf('webrtc://') === 0) {
+        const sdk = new SrsRtcPlayerAsync();
+        self.dom.prop('srcObject', sdk.stream);
+        sdk.play(self.url);
+        return console.log(`Play by srs.sdk.js for ${self.url}`);
+      }
+
+      console.error(`URL is not supported：${self.url}`);
+    };
+
+    return self;
+  };
+})(jQuery);
+

SrsPlayer/public/js/srs.sdk.js

@@ -0,0 +1,508 @@
+
+//
+// Copyright (c) 2013-2021 Winlin
+//
+// SPDX-License-Identifier: MIT
+//
+
+'use strict';
+
+function SrsError(name, message) {
+    this.name = name;
+    this.message = message;
+    this.stack = (new Error()).stack;
+}
+SrsError.prototype = Object.create(Error.prototype);
+SrsError.prototype.constructor = SrsError;
+
+// 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 = {};
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+    self.constraints = {
+        audio: true,
+        video: {
+            width: {ideal: 320, max: 576}
+        }
+    };
+
+    // @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(eip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?candidate=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"});
+
+        if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
+            throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
+        }
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+        stream.getTracks().forEach(function (track) {
+            self.pc.addTrack(track);
+
+            // Notify about local track when stream is ok.
+            self.ontrack && self.ontrack({track: 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);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200) return reject(xhr);
+                const data = JSON.parse(xhr.responseText);
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', conf.apiUrl, true);
+            xhr.setRequestHeader('Content-type', 'application/json');
+            xhr.send(JSON.stringify(data));
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+        );
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+        return session;
+    };
+
+    // Close the publisher.
+    self.close = function () {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got local stream.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    self.ontrack = function (event) {
+        // Add track to stream of SDK.
+        self.stream.addTrack(event.track);
+    };
+
+    // 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: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
+            };
+        },
+        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.substring(1, a.pathname.lastIndexOf("/"));
+            var stream = a.pathname.slice(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.slice(app.indexOf("?"));
+                app = app.slice(0, app.indexOf("?"));
+
+                if (params.indexOf("vhost=") > 0) {
+                    vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+                    if (vhost.indexOf("&") > 0) {
+                        vhost = vhost.slice(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+)$/;
+                if (re.test(a.hostname)) {
+                    vhost = "__defaultVhost__";
+                }
+            }
+
+            // parse the schema
+            var schema = "rtmp";
+            if (url.indexOf("://") > 0) {
+                schema = url.slice(0, url.indexOf("://"));
+            }
+
+            var port = a.port;
+            if (!port) {
+                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
+                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
+                    port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
+                }
+
+                // Guess by schema.
+                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);
+
+    // To keep api consistent between player and publisher.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    // @see https://webrtc.org/getting-started/media-devices
+    self.stream = new MediaStream();
+
+    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
+    //      webrtc://r.ossrs.net:80/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(eip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?candidate=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);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200) return reject(xhr);
+                const data = JSON.parse(xhr.responseText);
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', conf.apiUrl, true);
+            xhr.setRequestHeader('Content-type', 'application/json');
+            xhr.send(JSON.stringify(data));
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+        );
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+        return session;
+    };
+
+    // Close the player.
+    self.close = function() {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got remote track.
+    // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
+    self.ontrack = function (event) {
+        // https://webrtc.org/getting-started/remote-streams
+        self.stream.addTrack(event.track);
+    };
+
+    // 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: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
+            };
+        },
+        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.substring(1, a.pathname.lastIndexOf("/"));
+            var stream = a.pathname.slice(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.slice(app.indexOf("?"));
+                app = app.slice(0, app.indexOf("?"));
+
+                if (params.indexOf("vhost=") > 0) {
+                    vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+                    if (vhost.indexOf("&") > 0) {
+                        vhost = vhost.slice(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.slice(0, url.indexOf("://"));
+            }
+
+            var port = a.port;
+            if (!port) {
+                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
+                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
+                    port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
+                }
+
+                // Guess by schema.
+                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);
+
+    // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
+    self.stream = new MediaStream();
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
+    self.pc.ontrack = function(event) {
+        if (self.ontrack) {
+            self.ontrack(event);
+        }
+    };
+
+    return self;
+}
+