diff --git a/trunk/research/players/js/srs.page.js b/trunk/research/players/js/srs.page.js
index dbf4e7b6e..26b3971e1 100755
--- a/trunk/research/players/js/srs.page.js
+++ b/trunk/research/players/js/srs.page.js
@@ -1,6 +1,3 @@
-//////////////////////////////////////////////////////////////////////////////////
-//////////////////////////////////////////////////////////////////////////////////
-//////////////////////////////////////////////////////////////////////////////////
 
 // to query the swf anti cache.
 function srs_get_version_code() { return "1.33"; }
@@ -12,10 +9,6 @@ function srs_get_player_modal() { return 740; }
 function srs_get_player_width() { return srs_get_player_modal() - 30; }
 function srs_get_player_height() { return srs_get_player_width() * 9 / 19; }
 
-//////////////////////////////////////////////////////////////////////////////////
-//////////////////////////////////////////////////////////////////////////////////
-//////////////////////////////////////////////////////////////////////////////////
-
 /**
 * update the navigator, add same query string.
 */
@@ -93,34 +86,10 @@ function build_default_flv_url() {
 
     return uri;
 }
-// for the bandwidth tool to init page
-function build_default_bandwidth_rtmp_url() {
-    var query = parse_query_string();
-
-    var schema = 'rtmp';
-    var server = (!query.server)? window.location.hostname:query.server;
-    var port = (!query.port)? 1935:query.port;
-    var vhost = "bandcheck.srs.com";
-    var app = (!query.app)? "app":query.app;
-    var key = (!query.key)? "35c9b402c12a7246868752e2878f7e0e":query.key;
-
-    var uri = schema + "://" + server;
-    if (!is_default_port(schema, port)) {
-        uri += ":" + port;
-    }
-    uri += "/" + app + "?key=" + key + "&vhost=" + vhost;
-
-    return uri;
-}
 
 function build_default_rtc_url(query) {
-    // Use target to overwrite server, vhost and eip.
-    console.log('?target=x.x.x.x to overwrite server, vhost and eip.');
-    if (query.target) {
-        query.server = query.vhost = query.eip = query.target;
-        query.user_query.eip = query.target;
-        delete query.target;
-    }
+    // The format for query string to overwrite configs of server.
+    console.log('?eip=x.x.x.x to overwrite candidate. 覆盖服务器candidate(外网IP)配置');
 
     var server = (!query.server)? window.location.hostname:query.server;
     var vhost = (!query.vhost)? window.location.hostname:query.vhost;
@@ -165,253 +134,3 @@ function srs_init_rtc(id, query) {
     update_nav();
     $(id).val(build_default_rtc_url(query));
 }
-// for bw to init url
-// url: scheme://host:port/path?query#fragment
-function srs_init_bwt(rtmp_url, hls_url) {
-    update_nav();
-    
-    if (rtmp_url) {
-        $(rtmp_url).val(build_default_bandwidth_rtmp_url());
-    }
-}
-
-// check whether can republish
-function srs_can_republish() {
-    var browser = get_browser_agents();
-    
-    if (browser.Chrome || browser.Firefox) {
-        return true;
-    }
-    
-    if (browser.MSIE || browser.QQBrowser) {
-        return false;
-    }
-    
-    return false;
-}
-
-// without default values set.
-function srs_initialize_codec_page(
-    cameras, microphones,
-    sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
-    sl_acodec
-) {
-    $(sl_cameras).empty();
-    for (var i = 0; i < cameras.length; i++) {
-        $(sl_cameras).append("<option value='" + i + "'>" + cameras[i] + "</option");
-    }
-    // optional: select the except matches
-    matchs = ["virtual"];
-    for (var i = 0; i < cameras.length; i++) {
-        for (var j = 0; j < matchs.length; j++) {
-            if (cameras[i].toLowerCase().indexOf(matchs[j]) == -1) {
-                $(sl_cameras + " option[value='" + i + "']").attr("selected", true);
-                break;
-            }
-        }
-        if (j < matchs.length) {
-            break;
-        }
-    }
-    // optional: select the first matched.
-    matchs = ["truevision", "integrated"];
-    for (var i = 0; i < cameras.length; i++) {
-        for (var j = 0; j < matchs.length; j++) {
-            if (cameras[i].toLowerCase().indexOf(matchs[j]) >= 0) {
-                $(sl_cameras + " option[value='" + i + "']").attr("selected", true);
-                break;
-            }
-        }
-        if (j < matchs.length) {
-            break;
-        }
-    }
-    
-    $(sl_microphones).empty();
-    for (var i = 0; i < microphones.length; i++) {
-        $(sl_microphones).append("<option value='" + i + "'>" + microphones[i] + "</option");
-    }
-    // optional: select the except matches
-    matchs = ["default"];
-    for (var i = 0; i < microphones.length; i++) {
-        for (var j = 0; j < matchs.length; j++) {
-            if (microphones[i].toLowerCase().indexOf(matchs[j]) == -1) {
-                $(sl_microphones + " option[value='" + i + "']").attr("selected", true);
-                break;
-            }
-        }
-        if (j < matchs.length) {
-            break;
-        }
-    }
-    // optional: select the first matched.
-    matchs = ["realtek", "内置式麦克风"];
-    for (var i = 0; i < microphones.length; i++) {
-        for (var j = 0; j < matchs.length; j++) {
-            if (microphones[i].toLowerCase().indexOf(matchs[j]) >= 0) {
-                $(sl_microphones + " option[value='" + i + "']").attr("selected", true);
-                break;
-            }
-        }
-        if (j < matchs.length) {
-            break;
-        }
-    }
-    
-    $(sl_vcodec).empty();
-    var vcodecs = ["h264", "vp6"];
-    vcodecs = ["h264"]; // h264 only.
-    for (var i = 0; i < vcodecs.length; i++) {
-        $(sl_vcodec).append("<option value='" + vcodecs[i] + "'>" + vcodecs[i] + "</option");
-    }
-    
-    $(sl_profile).empty();
-    var profiles = ["baseline", "main"];
-    for (var i = 0; i < profiles.length; i++) {
-        $(sl_profile).append("<option value='" + profiles[i] + "'>" + profiles[i] + "</option");
-    }
-    
-    $(sl_level).empty();
-    var levels = ["1", "1b", "1.1", "1.2", "1.3", 
-        "2", "2.1", "2.2", "3", "3.1", "3.2", "4", "4.1", "4.2", "5", "5.1"];
-    for (var i = 0; i < levels.length; i++) {
-        $(sl_level).append("<option value='" + levels[i] + "'>" + levels[i] + "</option");
-    }
-    
-    $(sl_gop).empty();
-    var gops = ["0.3", "0.5", "1", "2", "3", "4", 
-        "5", "6", "7", "8", "9", "10", "15", "20"];
-    for (var i = 0; i < gops.length; i++) {
-        $(sl_gop).append("<option value='" + gops[i] + "'>" + gops[i] + "秒</option");
-    }
-    
-    $(sl_size).empty();
-    var sizes = ["176x144", "320x240", "352x240", 
-        "352x288", "480x360", "640x480", "720x480", "720x576", "800x600", 
-        "1024x768", "1280x720", "1360x768", "1920x1080"];
-    for (i = 0; i < sizes.length; i++) {
-        $(sl_size).append("<option value='" + sizes[i] + "'>" + sizes[i] + "</option");
-    }
-    
-    $(sl_fps).empty();
-    var fpses = ["5", "10", "15", "20", "24", "25", "29.97", "30"];
-    for (i = 0; i < fpses.length; i++) {
-        $(sl_fps).append("<option value='" + fpses[i] + "'>" + Number(fpses[i]).toFixed(2) + " 帧/秒</option");
-    }
-    
-    $(sl_bitrate).empty();
-    var bitrates = ["50", "200", "350", "500", "650", "800", 
-        "950", "1000", "1200", "1500", "1800", "2000", "3000", "5000"];
-    for (i = 0; i < bitrates.length; i++) {
-        $(sl_bitrate).append("<option value='" + bitrates[i] + "'>" + bitrates[i] + " kbps</option");
-    }
-    
-    $(sl_acodec).empty();
-    var bitrates = ["speex", "nellymoser", "pcma", "pcmu"];
-    for (i = 0; i < bitrates.length; i++) {
-        $(sl_acodec).append("<option value='" + bitrates[i] + "'>" + bitrates[i] + "</option");
-    }
-}
-/**
-* when publisher ready, init the page elements.
-*/
-function srs_publisher_initialize_page(
-    cameras, microphones,
-    sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
-    sl_acodec
-) {
-    srs_initialize_codec_page(
-        cameras, microphones,
-        sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
-        sl_acodec
-    );
-    
-    //var profiles = ["baseline", "main"];
-    $(sl_profile + " option[value='main']").attr("selected", true);
-    
-    //var levels = ["1", "1b", "1.1", "1.2", "1.3", 
-    //    "2", "2.1", "2.2", "3", "3.1", "3.2", "4", "4.1", "4.2", "5", "5.1"];
-    $(sl_level + " option[value='4.1']").attr("selected", true);
-    
-    //var gops = ["0.3", "0.5", "1", "2", "3", "4", 
-    //    "5", "6", "7", "8", "9", "10", "15", "20"];
-    $(sl_gop + " option[value='10']").attr("selected", true);
-    
-    //var sizes = ["176x144", "320x240", "352x240", 
-    //    "352x288", "480x360", "640x480", "720x480", "720x576", "800x600", 
-    //    "1024x768", "1280x720", "1360x768", "1920x1080"];
-    $(sl_size + " option[value='640x480']").attr("selected", true);
-    
-    //var fpses = ["5", "10", "15", "20", "24", "25", "29.97", "30"];
-    $(sl_fps + " option[value='20']").attr("selected", true);
-    
-    //var bitrates = ["50", "200", "350", "500", "650", "800", 
-    //    "950", "1000", "1200", "1500", "1800", "2000", "3000", "5000"];
-    $(sl_bitrate + " option[value='500']").attr("selected", true);
-    
-    // speex
-    $(sl_acodec + " option[value='speex']").attr("selected", true);
-}
-/**
-* for chat, use low latecy settings.
-*/
-function srs_chat_initialize_page(
-    cameras, microphones,
-    sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
-    sl_acodec
-) {
-    srs_initialize_codec_page(
-        cameras, microphones,
-        sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
-        sl_acodec
-    );
-    
-    //var profiles = ["baseline", "main"];
-    $(sl_profile + " option[value='baseline']").attr("selected", true);
-    
-    //var levels = ["1", "1b", "1.1", "1.2", "1.3", 
-    //    "2", "2.1", "2.2", "3", "3.1", "3.2", "4", "4.1", "4.2", "5", "5.1"];
-    $(sl_level + " option[value='3.1']").attr("selected", true);
-    
-    //var gops = ["0.3", "0.5", "1", "2", "3", "4", 
-    //    "5", "6", "7", "8", "9", "10", "15", "20"];
-    $(sl_gop + " option[value='2']").attr("selected", true);
-    
-    //var sizes = ["176x144", "320x240", "352x240", 
-    //    "352x288", "480x360", "640x480", "720x480", "720x576", "800x600", 
-    //    "1024x768", "1280x720", "1360x768", "1920x1080"];
-    $(sl_size + " option[value='480x360']").attr("selected", true);
-    
-    //var fpses = ["5", "10", "15", "20", "24", "25", "29.97", "30"];
-    $(sl_fps + " option[value='15']").attr("selected", true);
-    
-    //var bitrates = ["50", "200", "350", "500", "650", "800", 
-    //    "950", "1000", "1200", "1500", "1800", "2000", "3000", "5000"];
-    $(sl_bitrate + " option[value='350']").attr("selected", true);
-    
-    // speex
-    $(sl_acodec + " option[value='speex']").attr("selected", true);
-}
-/**
-* get the vcodec and acodec.
-*/
-function srs_publiser_get_codec(
-    vcodec, acodec,
-    sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
-    sl_acodec
-) {
-    acodec.codec       = $(sl_acodec).val();
-    acodec.device_code = $(sl_microphones).val();
-    acodec.device_name = $(sl_microphones).text();
-    
-    vcodec.device_code = $(sl_cameras).find("option:selected").val();
-    vcodec.device_name = $(sl_cameras).find("option:selected").text();
-    
-    vcodec.codec    = $(sl_vcodec).find("option:selected").val();
-    vcodec.profile  = $(sl_profile).find("option:selected").val();
-    vcodec.level    = $(sl_level).find("option:selected").val();
-    vcodec.fps      = $(sl_fps).find("option:selected").val();
-    vcodec.gop      = $(sl_gop).find("option:selected").val();
-    vcodec.size     = $(sl_size).find("option:selected").val();
-    vcodec.bitrate  = $(sl_bitrate).find("option:selected").val();
-}
diff --git a/trunk/research/players/js/srs.sdk.js b/trunk/research/players/js/srs.sdk.js
new file mode 100644
index 000000000..2b9f21feb
--- /dev/null
+++ b/trunk/research/players/js/srs.sdk.js
@@ -0,0 +1,472 @@
+
+/**
+ * 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, 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.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};
+        },
+        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, 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.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};
+        },
+        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;
+}
+
diff --git a/trunk/research/players/rtc_player.html b/trunk/research/players/rtc_player.html
index 1e7db005f..045bc8731 100644
--- a/trunk/research/players/rtc_player.html
+++ b/trunk/research/players/rtc_player.html
@@ -11,6 +11,7 @@
     <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/winlin.utility.js"></script>
     <script type="text/javascript" src="js/srs.page.js"></script>
 </head>
@@ -64,222 +65,6 @@
 </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();
diff --git a/trunk/research/players/rtc_publisher.html b/trunk/research/players/rtc_publisher.html
index fb62f0bf8..4eac03193 100644
--- a/trunk/research/players/rtc_publisher.html
+++ b/trunk/research/players/rtc_publisher.html
@@ -11,6 +11,7 @@
     <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/winlin.utility.js"></script>
     <script type="text/javascript" src="js/srs.page.js"></script>
 </head>
@@ -65,232 +66,6 @@
 <script type="text/javascript">
     var pc = null; // Global handler to do cleanup when replaying.
     $(function(){
-        // 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, 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.close();
-            };
-
-            // 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};
-                },
-                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;
-        }
-
         var sdk = null; // Global handler to do cleanup when republishing.
         var startPublish = function() {
             $('#rtc_media_player').show();
diff --git a/trunk/research/players/srs_bwt.html b/trunk/research/players/srs_bwt.html
index 3a96b62f2..a35d67e76 100644
--- a/trunk/research/players/srs_bwt.html
+++ b/trunk/research/players/srs_bwt.html
@@ -150,6 +150,35 @@
 <script type="text/javascript">
     var bandwidth = null;
 
+    // for bw to init url
+    // url: scheme://host:port/path?query#fragment
+    function srs_init_bwt(rtmp_url, hls_url) {
+        update_nav();
+
+        if (rtmp_url) {
+            $(rtmp_url).val(build_default_bandwidth_rtmp_url());
+        }
+    }
+    // for the bandwidth tool to init page
+    function build_default_bandwidth_rtmp_url() {
+        var query = parse_query_string();
+
+        var schema = 'rtmp';
+        var server = (!query.server)? window.location.hostname:query.server;
+        var port = (!query.port)? 1935:query.port;
+        var vhost = "bandcheck.srs.com";
+        var app = (!query.app)? "app":query.app;
+        var key = (!query.key)? "35c9b402c12a7246868752e2878f7e0e":query.key;
+
+        var uri = schema + "://" + server;
+        if (!is_default_port(schema, port)) {
+            uri += ":" + port;
+        }
+        uri += "/" + app + "?key=" + key + "&vhost=" + vhost;
+
+        return uri;
+    }
+
     var autoLoadPage = function() {
         srs_init_bwt("#txt_url");
 
diff --git a/trunk/research/players/srs_publisher_flash.html b/trunk/research/players/srs_publisher_flash.html
index c5de8c175..0010efc7c 100644
--- a/trunk/research/players/srs_publisher_flash.html
+++ b/trunk/research/players/srs_publisher_flash.html
@@ -438,6 +438,207 @@
             realtime_player.play(url);
         }
     }
+    /**
+     * get the vcodec and acodec.
+     */
+    function srs_publiser_get_codec(
+        vcodec, acodec,
+        sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
+        sl_acodec
+    ) {
+        acodec.codec       = $(sl_acodec).val();
+        acodec.device_code = $(sl_microphones).val();
+        acodec.device_name = $(sl_microphones).text();
+
+        vcodec.device_code = $(sl_cameras).find("option:selected").val();
+        vcodec.device_name = $(sl_cameras).find("option:selected").text();
+
+        vcodec.codec    = $(sl_vcodec).find("option:selected").val();
+        vcodec.profile  = $(sl_profile).find("option:selected").val();
+        vcodec.level    = $(sl_level).find("option:selected").val();
+        vcodec.fps      = $(sl_fps).find("option:selected").val();
+        vcodec.gop      = $(sl_gop).find("option:selected").val();
+        vcodec.size     = $(sl_size).find("option:selected").val();
+        vcodec.bitrate  = $(sl_bitrate).find("option:selected").val();
+    }
+    /**
+     * when publisher ready, init the page elements.
+     */
+    function srs_publisher_initialize_page(
+        cameras, microphones,
+        sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
+        sl_acodec
+    ) {
+        srs_initialize_codec_page(
+            cameras, microphones,
+            sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
+            sl_acodec
+        );
+
+        //var profiles = ["baseline", "main"];
+        $(sl_profile + " option[value='main']").attr("selected", true);
+
+        //var levels = ["1", "1b", "1.1", "1.2", "1.3",
+        //    "2", "2.1", "2.2", "3", "3.1", "3.2", "4", "4.1", "4.2", "5", "5.1"];
+        $(sl_level + " option[value='4.1']").attr("selected", true);
+
+        //var gops = ["0.3", "0.5", "1", "2", "3", "4",
+        //    "5", "6", "7", "8", "9", "10", "15", "20"];
+        $(sl_gop + " option[value='10']").attr("selected", true);
+
+        //var sizes = ["176x144", "320x240", "352x240",
+        //    "352x288", "480x360", "640x480", "720x480", "720x576", "800x600",
+        //    "1024x768", "1280x720", "1360x768", "1920x1080"];
+        $(sl_size + " option[value='640x480']").attr("selected", true);
+
+        //var fpses = ["5", "10", "15", "20", "24", "25", "29.97", "30"];
+        $(sl_fps + " option[value='20']").attr("selected", true);
+
+        //var bitrates = ["50", "200", "350", "500", "650", "800",
+        //    "950", "1000", "1200", "1500", "1800", "2000", "3000", "5000"];
+        $(sl_bitrate + " option[value='500']").attr("selected", true);
+
+        // speex
+        $(sl_acodec + " option[value='speex']").attr("selected", true);
+    }
+
+    // without default values set.
+    function srs_initialize_codec_page(
+        cameras, microphones,
+        sl_cameras, sl_microphones, sl_vcodec, sl_profile, sl_level, sl_gop, sl_size, sl_fps, sl_bitrate,
+        sl_acodec
+    ) {
+        $(sl_cameras).empty();
+        for (var i = 0; i < cameras.length; i++) {
+            $(sl_cameras).append("<option value='" + i + "'>" + cameras[i] + "</option");
+        }
+        // optional: select the except matches
+        matchs = ["virtual"];
+        for (var i = 0; i < cameras.length; i++) {
+            for (var j = 0; j < matchs.length; j++) {
+                if (cameras[i].toLowerCase().indexOf(matchs[j]) == -1) {
+                    $(sl_cameras + " option[value='" + i + "']").attr("selected", true);
+                    break;
+                }
+            }
+            if (j < matchs.length) {
+                break;
+            }
+        }
+        // optional: select the first matched.
+        matchs = ["truevision", "integrated"];
+        for (var i = 0; i < cameras.length; i++) {
+            for (var j = 0; j < matchs.length; j++) {
+                if (cameras[i].toLowerCase().indexOf(matchs[j]) >= 0) {
+                    $(sl_cameras + " option[value='" + i + "']").attr("selected", true);
+                    break;
+                }
+            }
+            if (j < matchs.length) {
+                break;
+            }
+        }
+
+        $(sl_microphones).empty();
+        for (var i = 0; i < microphones.length; i++) {
+            $(sl_microphones).append("<option value='" + i + "'>" + microphones[i] + "</option");
+        }
+        // optional: select the except matches
+        matchs = ["default"];
+        for (var i = 0; i < microphones.length; i++) {
+            for (var j = 0; j < matchs.length; j++) {
+                if (microphones[i].toLowerCase().indexOf(matchs[j]) == -1) {
+                    $(sl_microphones + " option[value='" + i + "']").attr("selected", true);
+                    break;
+                }
+            }
+            if (j < matchs.length) {
+                break;
+            }
+        }
+        // optional: select the first matched.
+        matchs = ["realtek", "内置式麦克风"];
+        for (var i = 0; i < microphones.length; i++) {
+            for (var j = 0; j < matchs.length; j++) {
+                if (microphones[i].toLowerCase().indexOf(matchs[j]) >= 0) {
+                    $(sl_microphones + " option[value='" + i + "']").attr("selected", true);
+                    break;
+                }
+            }
+            if (j < matchs.length) {
+                break;
+            }
+        }
+
+        $(sl_vcodec).empty();
+        var vcodecs = ["h264", "vp6"];
+        vcodecs = ["h264"]; // h264 only.
+        for (var i = 0; i < vcodecs.length; i++) {
+            $(sl_vcodec).append("<option value='" + vcodecs[i] + "'>" + vcodecs[i] + "</option");
+        }
+
+        $(sl_profile).empty();
+        var profiles = ["baseline", "main"];
+        for (var i = 0; i < profiles.length; i++) {
+            $(sl_profile).append("<option value='" + profiles[i] + "'>" + profiles[i] + "</option");
+        }
+
+        $(sl_level).empty();
+        var levels = ["1", "1b", "1.1", "1.2", "1.3",
+            "2", "2.1", "2.2", "3", "3.1", "3.2", "4", "4.1", "4.2", "5", "5.1"];
+        for (var i = 0; i < levels.length; i++) {
+            $(sl_level).append("<option value='" + levels[i] + "'>" + levels[i] + "</option");
+        }
+
+        $(sl_gop).empty();
+        var gops = ["0.3", "0.5", "1", "2", "3", "4",
+            "5", "6", "7", "8", "9", "10", "15", "20"];
+        for (var i = 0; i < gops.length; i++) {
+            $(sl_gop).append("<option value='" + gops[i] + "'>" + gops[i] + "秒</option");
+        }
+
+        $(sl_size).empty();
+        var sizes = ["176x144", "320x240", "352x240",
+            "352x288", "480x360", "640x480", "720x480", "720x576", "800x600",
+            "1024x768", "1280x720", "1360x768", "1920x1080"];
+        for (i = 0; i < sizes.length; i++) {
+            $(sl_size).append("<option value='" + sizes[i] + "'>" + sizes[i] + "</option");
+        }
+
+        $(sl_fps).empty();
+        var fpses = ["5", "10", "15", "20", "24", "25", "29.97", "30"];
+        for (i = 0; i < fpses.length; i++) {
+            $(sl_fps).append("<option value='" + fpses[i] + "'>" + Number(fpses[i]).toFixed(2) + " 帧/秒</option");
+        }
+
+        $(sl_bitrate).empty();
+        var bitrates = ["50", "200", "350", "500", "650", "800",
+            "950", "1000", "1200", "1500", "1800", "2000", "3000", "5000"];
+        for (i = 0; i < bitrates.length; i++) {
+            $(sl_bitrate).append("<option value='" + bitrates[i] + "'>" + bitrates[i] + " kbps</option");
+        }
+
+        $(sl_acodec).empty();
+        var bitrates = ["speex", "nellymoser", "pcma", "pcmu"];
+        for (i = 0; i < bitrates.length; i++) {
+            $(sl_acodec).append("<option value='" + bitrates[i] + "'>" + bitrates[i] + "</option");
+        }
+    }
+
+    // check whether can republish
+    function srs_can_republish() {
+        var browser = get_browser_agents();
+
+        if (browser.Chrome || browser.Firefox) {
+            return true;
+        }
+
+        if (browser.MSIE || browser.QQBrowser) {
+            return false;
+        }
+
+        return false;
+    }
 </script>
 </html>