From 0b62216999b4a20a5292263a50bc1526c8f556ca Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Fri, 30 Apr 2021 08:13:38 +0800
Subject: [PATCH] SquashSRS4: Support av1 for Chrome M90 enabled it.

---
 README.md                                     |   1 +
 trunk/research/players/js/srs.page.js         | 285 +---------
 trunk/research/players/js/srs.sdk.js          | 501 ++++++++++++++++++
 trunk/research/players/rtc_player.html        | 217 +-------
 trunk/research/players/rtc_publisher.html     | 239 +--------
 trunk/research/players/srs_bwt.html           |  29 +
 .../research/players/srs_publisher_flash.html | 201 +++++++
 trunk/scripts/git2unix.sh                     |  26 -
 trunk/src/app/srs_app_rtc_api.cpp             |  98 ++--
 trunk/src/app/srs_app_rtc_conn.cpp            | 247 ++++-----
 trunk/src/app/srs_app_rtc_conn.hpp            |  10 +-
 trunk/src/app/srs_app_rtc_server.cpp          |  58 +-
 trunk/src/app/srs_app_rtc_server.hpp          |  31 +-
 trunk/src/core/srs_core_version4.hpp          |   2 +-
 14 files changed, 980 insertions(+), 965 deletions(-)
 create mode 100644 trunk/research/players/js/srs.sdk.js
 delete mode 100755 trunk/scripts/git2unix.sh

diff --git a/README.md b/README.md
index 6e6daf59c..00820e063 100755
--- a/README.md
+++ b/README.md
@@ -163,6 +163,7 @@ Other important wiki:
 
 ## V4 changes
 
+* v4.0, 2021-04-29, RTC: Support av1 for Chrome M90. 4.0.91
 * v4.0, 2021-04-24, Change push-RTSP as deprecated feature.
 * v4.0, 2021-04-24, Player: Change the default from RTMP to HTTP-FLV.
 * v4.0, 2021-04-24, Disable CherryPy by --cherrypy=off. 4.0.90
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..6bde3fb8f
--- /dev/null
+++ b/trunk/research/players/js/srs.sdk.js
@@ -0,0 +1,501 @@
+
+/**
+ * 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;
+}
+
+// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
+// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
+function SrsRtcFormatSenders(senders, kind) {
+    var codecs = [];
+    senders.forEach(function (sender) {
+        sender.getParameters().codecs.forEach(function(c) {
+            if (kind && sender.track.kind !== kind) {
+                return;
+            }
+
+            if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
+                return;
+            }
+
+            var s = '';
+
+            s += c.mimeType.replace('audio/', '').replace('video/', '');
+            s += ', ' + c.clockRate + 'HZ';
+            if (sender.track.kind === "audio") {
+                s += ', channels: ' + c.channels;
+            }
+            s += ', pt: ' + c.payloadType;
+
+            codecs.push(s);
+        });
+    });
+    return codecs.join(", ");
+}
+
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..dda6e80f8 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>
@@ -54,6 +55,10 @@
     <label></label>
     SessionID: <span id='sessionid'></span>
 
+    <label></label>
+    Audio: <span id='acodecs'></span><br/>
+    Video: <span id='vcodecs'></span>
+
     <label></label>
     Simulator: <a href='#' id='simulator-drop'>Drop</a>
 
@@ -65,232 +70,6 @@
 <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();
@@ -306,6 +85,14 @@
                 $('#rtc_media_player').prop('srcObject', event.stream);
             };
 
+            // https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
+            sdk.pc.onicegatheringstatechange = function (event) {
+                if (sdk.pc.iceGatheringState === "complete") {
+                    $('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
+                    $('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
+                }
+            };
+
             // For example:
             //      webrtc://r.ossrs.net/live/livestream
             var url = $("#txt_url").val();
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>
 
diff --git a/trunk/scripts/git2unix.sh b/trunk/scripts/git2unix.sh
deleted file mode 100755
index 9aeddfb00..000000000
--- a/trunk/scripts/git2unix.sh
+++ /dev/null
@@ -1,26 +0,0 @@
-#!/bin/bash
-
-cat <<END >>/dev/null
-touch git2unix &&
-echo "bash `pwd`/git2unix.sh" >git2unix &&
-chmod +x git2unix &&
-sudo rm -f /bin/git2unix &&
-sudo mv git2unix /bin/git2unix
-END
-
-dos2unix -V>/dev/null 2>&1
-ret=$?; if [[ 0 -ne $ret ]]; then 
-    echo "dos2unix not found." 
-    echo "      sudo yum install -y dos2unix"
-    exit $ret
-fi
-
-files=`git status|egrep "(modified|new file)"|awk -F ':' '{print $2}'|awk '{print $1}'|egrep "(.hpp$|.cpp$|.cc$|.h$|.c$|.txt$|.sh|.conf$)"`;
-for file in $files; do
-    dos2unix $file;
-    echo $file|grep ".sh$" >/dev/null 2>&1; EOF_SH=$?
-    if [[ $EOF_SH -ne 0 && -f $file ]]; then 
-        echo "chmod -x $file"
-        chmod -x $file; 
-    fi
-done
diff --git a/trunk/src/app/srs_app_rtc_api.cpp b/trunk/src/app/srs_app_rtc_api.cpp
index fbe278b88..0916ec340 100644
--- a/trunk/src/app/srs_app_rtc_api.cpp
+++ b/trunk/src/app/srs_app_rtc_api.cpp
@@ -134,65 +134,69 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
 
     // For client to specifies the EIP of server.
     string eip = r->query_get("eip");
+    string codec = r->query_get("codec");
     // For client to specifies whether encrypt by SRTP.
     string srtp = r->query_get("encrypt");
     string dtls = r->query_get("dtls");
 
-    srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, srtp=%s, dtls=%s",
+    srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s",
         streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(),
-        srtp.c_str(), dtls.c_str());
+        codec.c_str(), srtp.c_str(), dtls.c_str()
+    );
+
+    // The RTC user config object.
+    SrsRtcUserConfig ruc;
+    ruc.eip_ = eip;
+    ruc.codec_ = codec;
+    ruc.publish_ = false;
+    ruc.dtls_ = (dtls != "false");
+
+    if (srtp.empty()) {
+        ruc.srtp_ = _srs_config->get_rtc_server_encrypt();
+    } else {
+        ruc.srtp_ = (srtp != "false");
+    }
 
     // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
-    SrsSdp remote_sdp;
-    if ((err = remote_sdp.parse(remote_sdp_str)) != srs_success) {
+    if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
         return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
     }
 
-    if ((err = check_remote_sdp(remote_sdp)) != srs_success) {
+    if ((err = check_remote_sdp(ruc.remote_sdp_)) != srs_success) {
         return srs_error_wrap(err, "remote sdp check failed");
     }
 
-    SrsRequest request;
-    request.app = app;
-    request.stream = stream_name;
+    ruc.req_->app = app;
+    ruc.req_->stream = stream_name;
 
     // TODO: FIXME: Parse vhost.
     // discovery vhost, resolve the vhost from config
     SrsConfDirective* parsed_vhost = _srs_config->get_vhost("");
     if (parsed_vhost) {
-        request.vhost = parsed_vhost->arg0();
+        ruc.req_->vhost = parsed_vhost->arg0();
     }
 
     SrsSdp local_sdp;
 
     // Config for SDP and session.
-    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(request.vhost);
-    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(request.vhost);
+    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost);
+    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost);
 
     // Whether enabled.
     bool server_enabled = _srs_config->get_rtc_server_enabled();
-    bool rtc_enabled = _srs_config->get_rtc_enabled(request.vhost);
+    bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost);
     if (server_enabled && !rtc_enabled) {
-        srs_warn("RTC disabled in vhost %s", request.vhost.c_str());
+        srs_warn("RTC disabled in vhost %s", ruc.req_->vhost.c_str());
     }
     if (!server_enabled || !rtc_enabled) {
         return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
-            server_enabled, rtc_enabled, request.vhost.c_str());
+            server_enabled, rtc_enabled, ruc.req_->vhost.c_str());
     }
 
-    bool srtp_enabled = true;
-    if (srtp.empty()) {
-        srtp_enabled = _srs_config->get_rtc_server_encrypt();
-    } else {
-        srtp_enabled = (srtp != "false");
-    }
-
-    bool dtls_enabled = (dtls != "false");
-
     // TODO: FIXME: When server enabled, but vhost disabled, should report error.
     SrsRtcConnection* session = NULL;
-    if ((err = server_->create_session(&request, remote_sdp, local_sdp, eip, false, dtls_enabled, srtp_enabled, &session)) != srs_success) {
-        return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", dtls_enabled, srtp_enabled, eip.c_str());
+    if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) {
+        return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", ruc.dtls_, ruc.srtp_, eip.c_str());
     }
 
     ostringstream os;
@@ -213,7 +217,7 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
     res->set("sessionid", SrsJsonAny::str(session->username().c_str()));
 
     srs_trace("RTC username=%s, dtls=%u, srtp=%u, offer=%dB, answer=%dB", session->username().c_str(),
-        dtls_enabled, srtp_enabled, remote_sdp_str.length(), local_sdp_str.length());
+        ruc.dtls_, ruc.srtp_, remote_sdp_str.length(), local_sdp_str.length());
     srs_trace("RTC remote offer: %s", srs_string_replace(remote_sdp_str.c_str(), "\r\n", "\\r\\n").c_str());
     srs_trace("RTC local answer: %s", local_sdp_str.c_str());
 
@@ -301,7 +305,7 @@ srs_error_t SrsGoApiRtcPlay::exchange_sdp(SrsRequest* req, const SrsSdp& remote_
                     }
                 }
 
-                // Only choose one match opus codec.
+                // Only choose one match opus.
                 break;
             }
 
@@ -498,52 +502,60 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
 
     // For client to specifies the EIP of server.
     string eip = r->query_get("eip");
+    string codec = r->query_get("codec");
 
-    srs_trace("RTC publish %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s",
-        streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str());
+    srs_trace("RTC publish %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
+        streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(),
+        codec.c_str()
+    );
+
+    // The RTC user config object.
+    SrsRtcUserConfig ruc;
+    ruc.eip_ = eip;
+    ruc.codec_ = codec;
+    ruc.publish_ = true;
+    ruc.dtls_ = ruc.srtp_ = true;
 
     // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
-    SrsSdp remote_sdp;
-    if ((err = remote_sdp.parse(remote_sdp_str)) != srs_success) {
+    if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
         return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
     }
 
-    if ((err = check_remote_sdp(remote_sdp)) != srs_success) {
+    if ((err = check_remote_sdp(ruc.remote_sdp_)) != srs_success) {
         return srs_error_wrap(err, "remote sdp check failed");
     }
 
-    SrsRequest request;
-    request.app = app;
-    request.stream = stream_name;
+    ruc.req_->app = app;
+    ruc.req_->stream = stream_name;
 
     // TODO: FIXME: Parse vhost.
     // discovery vhost, resolve the vhost from config
     SrsConfDirective* parsed_vhost = _srs_config->get_vhost("");
     if (parsed_vhost) {
-        request.vhost = parsed_vhost->arg0();
+        ruc.req_->vhost = parsed_vhost->arg0();
     }
 
     SrsSdp local_sdp;
 
     // TODO: FIXME: move to create_session.
     // Config for SDP and session.
-    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(request.vhost);
-    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(request.vhost);
+    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost);
+    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost);
 
     // Whether enabled.
     bool server_enabled = _srs_config->get_rtc_server_enabled();
-    bool rtc_enabled = _srs_config->get_rtc_enabled(request.vhost);
+    bool rtc_enabled = _srs_config->get_rtc_enabled(ruc.req_->vhost);
     if (server_enabled && !rtc_enabled) {
-        srs_warn("RTC disabled in vhost %s", request.vhost.c_str());
+        srs_warn("RTC disabled in vhost %s", ruc.req_->vhost.c_str());
     }
     if (!server_enabled || !rtc_enabled) {
         return srs_error_new(ERROR_RTC_DISABLED, "Disabled server=%d, rtc=%d, vhost=%s",
-            server_enabled, rtc_enabled, request.vhost.c_str());
+            server_enabled, rtc_enabled, ruc.req_->vhost.c_str());
     }
 
     // TODO: FIXME: When server enabled, but vhost disabled, should report error.
     SrsRtcConnection* session = NULL;
-    if ((err = server_->create_session(&request, remote_sdp, local_sdp, eip, true, true, true, &session)) != srs_success) {
+    if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) {
         return srs_error_wrap(err, "create session");
     }
 
@@ -674,7 +686,7 @@ srs_error_t SrsGoApiRtcPublish::exchange_sdp(SrsRequest* req, const SrsSdp& remo
                     }
                 }
 
-                // Only choose one match opus codec.
+                // Only choose one match opus.
                 break;
             }
 
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index 2aecf96e7..2716ddd44 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -1806,19 +1806,21 @@ const SrsContextId& SrsRtcConnection::context_id()
     return cid_;
 }
 
-srs_error_t SrsRtcConnection::add_publisher(SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp)
+srs_error_t SrsRtcConnection::add_publisher(SrsRtcUserConfig* ruc, SrsSdp& local_sdp)
 {
     srs_error_t err = srs_success;
 
+    SrsRequest* req = ruc->req_;
+
     SrsRtcStreamDescription* stream_desc = new SrsRtcStreamDescription();
     SrsAutoFree(SrsRtcStreamDescription, stream_desc);
 
     // TODO: FIXME: Change to api of stream desc.
-    if ((err = negotiate_publish_capability(req, remote_sdp, stream_desc)) != srs_success) {
+    if ((err = negotiate_publish_capability(ruc, stream_desc)) != srs_success) {
         return srs_error_wrap(err, "publish negotiate");
     }
 
-    if ((err = generate_publish_local_sdp(req, local_sdp, stream_desc, remote_sdp.is_unified())) != srs_success) {
+    if ((err = generate_publish_local_sdp(req, local_sdp, stream_desc, ruc->remote_sdp_.is_unified())) != srs_success) {
         return srs_error_wrap(err, "generate local sdp");
     }
 
@@ -1846,10 +1848,12 @@ srs_error_t SrsRtcConnection::add_publisher(SrsRequest* req, const SrsSdp& remot
 }
 
 // TODO: FIXME: Error when play before publishing.
-srs_error_t SrsRtcConnection::add_player(SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp)
+srs_error_t SrsRtcConnection::add_player(SrsRtcUserConfig* ruc, SrsSdp& local_sdp)
 {
     srs_error_t err = srs_success;
 
+    SrsRequest* req = ruc->req_;
+
     if (_srs_rtc_hijacker) {
         if ((err = _srs_rtc_hijacker->on_before_play(this, req)) != srs_success) {
             return srs_error_wrap(err, "before play");
@@ -1857,7 +1861,7 @@ srs_error_t SrsRtcConnection::add_player(SrsRequest* req, const SrsSdp& remote_s
     }
 
     std::map<uint32_t, SrsRtcTrackDescription*> play_sub_relations;
-    if ((err = negotiate_play_capability(req, remote_sdp, play_sub_relations)) != srs_success) {
+    if ((err = negotiate_play_capability(ruc, play_sub_relations)) != srs_success) {
         return srs_error_wrap(err, "play negotiate");
     }
 
@@ -1882,7 +1886,7 @@ srs_error_t SrsRtcConnection::add_player(SrsRequest* req, const SrsSdp& remote_s
         ++it;
     }
 
-    if ((err = generate_play_local_sdp(req, local_sdp, stream_desc, remote_sdp.is_unified())) != srs_success) {
+    if ((err = generate_play_local_sdp(req, local_sdp, stream_desc, ruc->remote_sdp_.is_unified())) != srs_success) {
         return srs_error_wrap(err, "generate local sdp");
     }
 
@@ -2667,7 +2671,7 @@ bool srs_sdp_has_h264_profile(const SrsSdp& sdp, const string& profile)
     return false;
 }
 
-srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, const SrsSdp& remote_sdp, SrsRtcStreamDescription* stream_desc)
+srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc, SrsRtcStreamDescription* stream_desc)
 {
     srs_error_t err = srs_success;
 
@@ -2675,13 +2679,16 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
         return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "stream description is NULL");
     }
 
+    SrsRequest* req = ruc->req_;
+    const SrsSdp& remote_sdp = ruc->remote_sdp_;
+
     bool nack_enabled = _srs_config->get_rtc_nack_enabled(req->vhost);
     bool twcc_enabled = _srs_config->get_rtc_twcc_enabled(req->vhost);
     // TODO: FIME: Should check packetization-mode=1 also.
     bool has_42e01f = srs_sdp_has_h264_profile(remote_sdp, "42e01f");
 
-    for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
-        const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_[i];
+    for (int i = 0; i < (int)remote_sdp.media_descs_.size(); ++i) {
+        const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_.at(i);
 
         SrsRtcTrackDescription* track_desc = new SrsRtcTrackDescription();
         SrsAutoFree(SrsRtcTrackDescription, track_desc);
@@ -2711,28 +2718,66 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
                 return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found opus payload type");
             }
 
-            for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
-                // if the playload is opus, and the encoding_param_ is channel
-                SrsAudioPayload* audio_payload = new SrsAudioPayload(iter->payload_type_, iter->encoding_name_, iter->clock_rate_, ::atol(iter->encoding_param_.c_str()));
-                audio_payload->set_opus_param_desc(iter->format_specific_param_);
+            for (int j = 0; j < (int)payloads.size(); j++) {
+                const SrsMediaPayloadType& payload = payloads.at(j);
+
+                // if the payload is opus, and the encoding_param_ is channel
+                SrsAudioPayload* audio_payload = new SrsAudioPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_, ::atol(payload.encoding_param_.c_str()));
+                audio_payload->set_opus_param_desc(payload.format_specific_param_);
+
                 // TODO: FIXME: Only support some transport algorithms.
-                for (int j = 0; j < (int)iter->rtcp_fb_.size(); ++j) {
+                for (int k = 0; k < (int)payload.rtcp_fb_.size(); ++k) {
+                    const string& rtcp_fb = payload.rtcp_fb_.at(k);
+
                     if (nack_enabled) {
-                        if (iter->rtcp_fb_.at(j) == "nack" || iter->rtcp_fb_.at(j) == "nack pli") {
-                            audio_payload->rtcp_fbs_.push_back(iter->rtcp_fb_.at(j));
+                        if (rtcp_fb == "nack" || rtcp_fb == "nack pli") {
+                            audio_payload->rtcp_fbs_.push_back(rtcp_fb);
                         }
                     }
                     if (twcc_enabled && remote_twcc_id) {
-                        if (iter->rtcp_fb_.at(j) == "transport-cc") {
-                            audio_payload->rtcp_fbs_.push_back(iter->rtcp_fb_.at(j));
+                        if (rtcp_fb == "transport-cc") {
+                            audio_payload->rtcp_fbs_.push_back(rtcp_fb);
                         }
                     }
                 }
+
                 track_desc->type_ = "audio";
                 track_desc->set_codec_payload((SrsCodecPayload*)audio_payload);
                 // Only choose one match opus codec.
                 break;
             }
+        } else if (remote_media_desc.is_video() && ruc->codec_ == "av1") {
+            std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("AV1X");
+            if (payloads.empty()) {
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid AV1 payload type");
+            }
+
+            for (int j = 0; j < (int)payloads.size(); j++) {
+                const SrsMediaPayloadType& payload = payloads.at(j);
+
+                // Generate video payload for av1.
+                SrsVideoPayload* video_payload = new SrsVideoPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_);
+
+                // TODO: FIXME: Only support some transport algorithms.
+                for (int k = 0; k < (int)payload.rtcp_fb_.size(); ++k) {
+                    const string& rtcp_fb = payload.rtcp_fb_.at(k);
+
+                    if (nack_enabled) {
+                        if (rtcp_fb == "nack" || rtcp_fb == "nack pli") {
+                            video_payload->rtcp_fbs_.push_back(rtcp_fb);
+                        }
+                    }
+                    if (twcc_enabled && remote_twcc_id) {
+                        if (rtcp_fb == "transport-cc") {
+                            video_payload->rtcp_fbs_.push_back(rtcp_fb);
+                        }
+                    }
+                }
+
+                track_desc->type_ = "video";
+                track_desc->set_codec_payload((SrsCodecPayload*)video_payload);
+                break;
+            }
         } else if (remote_media_desc.is_video()) {
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
             if (payloads.empty()) {
@@ -2740,13 +2785,15 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
             }
 
             std::deque<SrsMediaPayloadType> backup_payloads;
-            for (std::vector<SrsMediaPayloadType>::iterator iter = payloads.begin(); iter != payloads.end(); ++iter) {
-                if (iter->format_specific_param_.empty()) {
-                    backup_payloads.push_front(*iter);
+            for (int j = 0; j < (int)payloads.size(); j++) {
+                const SrsMediaPayloadType& payload = payloads.at(j);
+
+                if (payload.format_specific_param_.empty()) {
+                    backup_payloads.push_front(payload);
                     continue;
                 }
                 H264SpecificParam h264_param;
-                if ((err = srs_parse_h264_fmtp(iter->format_specific_param_, h264_param)) != srs_success) {
+                if ((err = srs_parse_h264_fmtp(payload.format_specific_param_, h264_param)) != srs_success) {
                     srs_error_reset(err); continue;
                 }
 
@@ -2754,21 +2801,23 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
                 bool profile_matched = (!has_42e01f || h264_param.profile_level_id == "42e01f");
 
                 // Try to pick the "best match" H.264 payload type.
-                if (h264_param.packetization_mode == "1" && h264_param.level_asymmerty_allow == "1" && profile_matched) {
+                if (profile_matched && h264_param.packetization_mode == "1" && h264_param.level_asymmerty_allow == "1") {
                     // if the playload is opus, and the encoding_param_ is channel
-                    SrsVideoPayload* video_payload = new SrsVideoPayload(iter->payload_type_, iter->encoding_name_, iter->clock_rate_);
-                    video_payload->set_h264_param_desc(iter->format_specific_param_);
+                    SrsVideoPayload* video_payload = new SrsVideoPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_);
+                    video_payload->set_h264_param_desc(payload.format_specific_param_);
 
                     // TODO: FIXME: Only support some transport algorithms.
-                    for (int j = 0; j < (int)iter->rtcp_fb_.size(); ++j) {
+                    for (int k = 0; k < (int)payload.rtcp_fb_.size(); ++k) {
+                        const string& rtcp_fb = payload.rtcp_fb_.at(k);
+
                         if (nack_enabled) {
-                            if (iter->rtcp_fb_.at(j) == "nack" || iter->rtcp_fb_.at(j) == "nack pli") {
-                                video_payload->rtcp_fbs_.push_back(iter->rtcp_fb_.at(j));
+                            if (rtcp_fb == "nack" || rtcp_fb == "nack pli") {
+                                video_payload->rtcp_fbs_.push_back(rtcp_fb);
                             }
                         }
                         if (twcc_enabled && remote_twcc_id) {
-                            if (iter->rtcp_fb_.at(j) == "transport-cc") {
-                                video_payload->rtcp_fbs_.push_back(iter->rtcp_fb_.at(j));
+                            if (rtcp_fb == "transport-cc") {
+                                video_payload->rtcp_fbs_.push_back(rtcp_fb);
                             }
                         }
                     }
@@ -2779,34 +2828,35 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
                     break;
                 }
 
-                backup_payloads.push_back(*iter);
+                backup_payloads.push_back(payload);
             }
 
             // Try my best to pick at least one media payload type.
             if (!track_desc->media_ && ! backup_payloads.empty()) {
-                SrsMediaPayloadType media_pt= backup_payloads.front();
-                // if the playload is opus, and the encoding_param_ is channel
-                SrsVideoPayload* video_payload = new SrsVideoPayload(media_pt.payload_type_, media_pt.encoding_name_, media_pt.clock_rate_);
+                const SrsMediaPayloadType& payload = backup_payloads.front();
+
+                // if the playload is opus, and the encoding_param_ is channel
+                SrsVideoPayload* video_payload = new SrsVideoPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_);
 
-                std::vector<std::string> rtcp_fbs = media_pt.rtcp_fb_;
                 // TODO: FIXME: Only support some transport algorithms.
-                for (int j = 0; j < (int)rtcp_fbs.size(); ++j) {
+                for (int k = 0; k < (int)payload.rtcp_fb_.size(); ++k) {
+                    const string& rtcp_fb = payload.rtcp_fb_.at(k);
+
                     if (nack_enabled) {
-                        if (rtcp_fbs.at(j) == "nack" || rtcp_fbs.at(j) == "nack pli") {
-                            video_payload->rtcp_fbs_.push_back(rtcp_fbs.at(j));
+                        if (rtcp_fb == "nack" || rtcp_fb == "nack pli") {
+                            video_payload->rtcp_fbs_.push_back(rtcp_fb);
                         }
                     }
 
                     if (twcc_enabled && remote_twcc_id) {
-                        if (rtcp_fbs.at(j) == "transport-cc") {
-                            video_payload->rtcp_fbs_.push_back(rtcp_fbs.at(j));
+                        if (rtcp_fb == "transport-cc") {
+                            video_payload->rtcp_fbs_.push_back(rtcp_fb);
                         }
                     }
                 }
 
                 track_desc->set_codec_payload((SrsCodecPayload*)video_payload);
-
-                srs_warn("choose backup H.264 payload type=%d", backup_payloads.front().payload_type_);
+                srs_warn("choose backup H.264 payload type=%d", payload.payload_type_);
             }
 
             // TODO: FIXME: Support RRTR?
@@ -2820,8 +2870,9 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
         track_desc->create_auxiliary_payload(remote_media_desc.find_media_with_encoding_name("ulpfec"));
 
         std::string track_id;
-        for (int i = 0; i < (int)remote_media_desc.ssrc_infos_.size(); ++i) {
-            SrsSSRCInfo ssrc_info = remote_media_desc.ssrc_infos_.at(i);
+        for (int j = 0; j < (int)remote_media_desc.ssrc_infos_.size(); ++j) {
+            const SrsSSRCInfo& ssrc_info = remote_media_desc.ssrc_infos_.at(j);
+
             // ssrc have same track id, will be description in the same track description.
             if(track_id != ssrc_info.msid_tracker_) {
                 SrsRtcTrackDescription* track_desc_copy = track_desc->copy();
@@ -2839,8 +2890,9 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRequest* req, cons
         }
 
         // set track fec_ssrc and rtx_ssrc
-        for (int i = 0; i < (int)remote_media_desc.ssrc_groups_.size(); ++i) {
-            SrsSSRCGroup ssrc_group = remote_media_desc.ssrc_groups_.at(i);
+        for (int j = 0; j < (int)remote_media_desc.ssrc_groups_.size(); ++j) {
+            const SrsSSRCGroup& ssrc_group = remote_media_desc.ssrc_groups_.at(j);
+
             SrsRtcTrackDescription* track_desc = stream_desc->find_track_description_by_ssrc(ssrc_group.ssrcs_[0]);
             if (!track_desc) {
                 continue;
@@ -2962,10 +3014,13 @@ srs_error_t SrsRtcConnection::generate_publish_local_sdp(SrsRequest* req, SrsSdp
     return err;
 }
 
-srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const SrsSdp& remote_sdp, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations)
+srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRtcUserConfig* ruc, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations)
 {
     srs_error_t err = srs_success;
 
+    SrsRequest* req = ruc->req_;
+    const SrsSdp& remote_sdp = ruc->remote_sdp_;
+
     bool nack_enabled = _srs_config->get_rtc_nack_enabled(req->vhost);
     bool twcc_enabled = _srs_config->get_rtc_twcc_enabled(req->vhost);
     // TODO: FIME: Should check packetization-mode=1 also.
@@ -2976,8 +3031,8 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
         return srs_error_wrap(err, "fetch rtc source");
     }
 
-    for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
-        const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_[i];
+    for (int i = 0; i < (int)remote_sdp.media_descs_.size(); ++i) {
+        const SrsMediaDesc& remote_media_desc = remote_sdp.media_descs_.at(i);
 
         // Whether feature enabled in remote extmap.
         int remote_twcc_id = 0;
@@ -3002,6 +3057,14 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
 
             remote_payload = payloads.at(0);
             track_descs = source->get_track_desc("audio", "opus");
+        } else if (remote_media_desc.is_video() && ruc->codec_ == "av1") {
+            std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("AV1X");
+            if (payloads.empty()) {
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid AV1 payload type");
+            }
+
+            remote_payload = payloads.at(0);
+            track_descs = source->get_track_desc("video", "AV1X");
         } else if (remote_media_desc.is_video()) {
             // TODO: check opus format specific param
             vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
@@ -3011,7 +3074,7 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
 
             remote_payload = payloads.at(0);
             for (int j = 0; j < (int)payloads.size(); j++) {
-                SrsMediaPayloadType& payload = payloads.at(j);
+                const SrsMediaPayloadType& payload = payloads.at(j);
 
                 // If exists 42e01f profile, choose it; otherwise, use the first payload.
                 // TODO: FIME: Should check packetization-mode=1 also.
@@ -3024,8 +3087,8 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
             track_descs = source->get_track_desc("video", "H264");
         }
 
-        for (int i = 0; i < (int)track_descs.size(); ++i) {
-            SrsRtcTrackDescription* track = track_descs[i]->copy();
+        for (int j = 0; j < (int)track_descs.size(); ++j) {
+            SrsRtcTrackDescription* track = track_descs.at(j)->copy();
 
             // Use remote/source/offer PayloadType.
             track->media_->pt_of_publisher_ = track->media_->pt_;
@@ -3077,86 +3140,6 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, const S
     return err;
 }
 
-srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRequest* req, SrsRtcStreamDescription* req_stream_desc,
-        std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations)
-{
-    srs_error_t err = srs_success;
-
-    SrsRtcStream* source = NULL;
-    if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
-        return srs_error_wrap(err, "fetch rtc source");
-    }
-
-    std::vector<SrsRtcTrackDescription*> src_track_descs;
-    //negotiate audio media
-    if(NULL != req_stream_desc->audio_track_desc_) {
-        SrsRtcTrackDescription* req_audio_track = req_stream_desc->audio_track_desc_;
-        int remote_twcc_id = req_audio_track->get_rtp_extension_id(kTWCCExt);
-
-        src_track_descs = source->get_track_desc("audio", "opus");
-        if (src_track_descs.size() > 0) {
-            // FIXME: use source sdp or subscribe sdp? native subscribe may have no sdp
-            SrsRtcTrackDescription *track = src_track_descs[0]->copy();
-
-            // Use remote/source/offer PayloadType.
-            track->media_->pt_of_publisher_ = track->media_->pt_;
-            track->media_->pt_ = req_audio_track->media_->pt_;
-
-            if (req_audio_track->red_ && track->red_) {
-                track->red_->pt_of_publisher_ = track->red_->pt_;
-                track->red_->pt_ = req_audio_track->red_->pt_;
-            }
-
-            track->del_rtp_extension_desc(kTWCCExt);
-            if (remote_twcc_id > 0) {
-                track->add_rtp_extension_desc(remote_twcc_id, kTWCCExt);
-            }
-
-            track->mid_ = req_audio_track->mid_;
-            sub_relations.insert(make_pair(track->ssrc_, track));
-            track->set_direction("sendonly");
-            track->ssrc_ = SrsRtcSSRCGenerator::instance()->generate_ssrc();
-        }
-    }
-
-    //negotiate video media
-    std::vector<SrsRtcTrackDescription*> req_video_tracks = req_stream_desc->video_track_descs_;
-    src_track_descs = source->get_track_desc("video", "h264");
-    for(int i = 0; i < (int)req_video_tracks.size(); ++i) {
-        SrsRtcTrackDescription* req_video = req_video_tracks.at(i);
-        int remote_twcc_id = req_video->get_rtp_extension_id(kTWCCExt);
-
-        for(int j = 0; j < (int)src_track_descs.size(); ++j) {
-            SrsRtcTrackDescription* src_video = src_track_descs.at(j);
-            if(req_video->id_ == src_video->id_) {
-                // FIXME: use source sdp or subscribe sdp? native subscribe may have no sdp
-                SrsRtcTrackDescription *track = src_video->copy();
-
-                // Use remote/source/offer PayloadType.
-                track->media_->pt_of_publisher_ = track->media_->pt_;
-                track->media_->pt_ = req_video->media_->pt_;
-
-                if (req_video->red_ && track->red_) {
-                    track->red_->pt_of_publisher_ = track->red_->pt_;
-                    track->red_->pt_ = req_video->red_->pt_;
-                }
-
-                track->del_rtp_extension_desc(kTWCCExt);
-                if (remote_twcc_id > 0) {
-                    track->add_rtp_extension_desc(remote_twcc_id, kTWCCExt);
-                }
-
-                track->mid_ = req_video->mid_;
-                sub_relations.insert(make_pair(track->ssrc_, track));
-                track->set_direction("sendonly");
-                track->ssrc_ = SrsRtcSSRCGenerator::instance()->generate_ssrc();
-            }
-        }
-    }
-
-    return err;
-}
-
 srs_error_t SrsRtcConnection::fetch_source_capability(SrsRequest* req, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations)
 {
     srs_error_t err = srs_success;
diff --git a/trunk/src/app/srs_app_rtc_conn.hpp b/trunk/src/app/srs_app_rtc_conn.hpp
index 694f63c73..0f219f58b 100644
--- a/trunk/src/app/srs_app_rtc_conn.hpp
+++ b/trunk/src/app/srs_app_rtc_conn.hpp
@@ -64,6 +64,7 @@ class SrsRtcVideoSendTrack;
 class SrsErrorPithyPrint;
 class SrsPithyPrint;
 class SrsStatistic;
+class SrsRtcUserConfig;
 
 const uint8_t kSR   = 200;
 const uint8_t kRR   = 201;
@@ -476,8 +477,8 @@ public:
     void switch_to_context();
     const SrsContextId& context_id();
 public:
-    srs_error_t add_publisher(SrsRequest* request, const SrsSdp& remote_sdp, SrsSdp& local_sdp);
-    srs_error_t add_player(SrsRequest* request, const SrsSdp& remote_sdp, SrsSdp& local_sdp);
+    srs_error_t add_publisher(SrsRtcUserConfig* ruc, SrsSdp& local_sdp);
+    srs_error_t add_player(SrsRtcUserConfig* ruc, SrsSdp& local_sdp);
 public:
     // Before initialize, user must set the local SDP, which is used to inititlize DTLS.
     srs_error_t initialize(SrsRequest* r, bool dtls, bool srtp, std::string username);
@@ -525,12 +526,11 @@ public:
 private:
     srs_error_t on_binding_request(SrsStunPacket* r);
     // publish media capabilitiy negotiate
-    srs_error_t negotiate_publish_capability(SrsRequest* req, const SrsSdp& remote_sdp, SrsRtcStreamDescription* stream_desc);
+    srs_error_t negotiate_publish_capability(SrsRtcUserConfig* ruc, SrsRtcStreamDescription* stream_desc);
     srs_error_t generate_publish_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcStreamDescription* stream_desc, bool unified_plan);
     // play media capabilitiy negotiate
     //TODO: Use StreamDescription to negotiate and remove first negotiate_play_capability function
-    srs_error_t negotiate_play_capability(SrsRequest* req, const SrsSdp& remote_sdp, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations);
-    srs_error_t negotiate_play_capability(SrsRequest* req, SrsRtcStreamDescription* req_stream_desc, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations);
+    srs_error_t negotiate_play_capability(SrsRtcUserConfig* ruc, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations);
     srs_error_t fetch_source_capability(SrsRequest* req, std::map<uint32_t, SrsRtcTrackDescription*>& sub_relations);
     srs_error_t generate_play_local_sdp(SrsRequest* req, SrsSdp& local_sdp, SrsRtcStreamDescription* stream_desc, bool unified_plan);
     srs_error_t create_player(SrsRequest* request, std::map<uint32_t, SrsRtcTrackDescription*> sub_relations);
diff --git a/trunk/src/app/srs_app_rtc_server.cpp b/trunk/src/app/srs_app_rtc_server.cpp
index 4ba4e6ff4..0f83bc9a6 100644
--- a/trunk/src/app/srs_app_rtc_server.cpp
+++ b/trunk/src/app/srs_app_rtc_server.cpp
@@ -243,6 +243,18 @@ ISrsRtcServerHijacker::~ISrsRtcServerHijacker()
 {
 }
 
+SrsRtcUserConfig::SrsRtcUserConfig()
+{
+    req_ = new SrsRequest();
+    publish_ = false;
+    dtls_ = srtp_ = true;
+}
+
+SrsRtcUserConfig::~SrsRtcUserConfig()
+{
+    srs_freep(req_);
+}
+
 SrsRtcServer::SrsRtcServer()
 {
     handler = NULL;
@@ -498,27 +510,26 @@ srs_error_t SrsRtcServer::listen_api()
     return err;
 }
 
-srs_error_t SrsRtcServer::create_session(
-    SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp, const std::string& mock_eip,
-    bool publish, bool dtls, bool srtp,
-    SrsRtcConnection** psession
-) {
+srs_error_t SrsRtcServer::create_session(SrsRtcUserConfig* ruc, SrsSdp& local_sdp, SrsRtcConnection** psession)
+{
     srs_error_t err = srs_success;
 
     SrsContextId cid = _srs_context->get_id();
 
+    SrsRequest* req = ruc->req_;
+
     SrsRtcStream* source = NULL;
     if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
         return srs_error_wrap(err, "create source");
     }
 
-    if (publish && !source->can_publish()) {
+    if (ruc->publish_ && !source->can_publish()) {
         return srs_error_new(ERROR_RTC_SOURCE_BUSY, "stream %s busy", req->get_stream_url().c_str());
     }
 
     // TODO: FIXME: add do_create_session to error process.
     SrsRtcConnection* session = new SrsRtcConnection(this, cid);
-    if ((err = do_create_session(session, req, remote_sdp, local_sdp, mock_eip, publish, dtls, srtp)) != srs_success) {
+    if ((err = do_create_session(ruc, local_sdp, session)) != srs_success) {
         srs_freep(session);
         return srs_error_wrap(err, "create session");
     }
@@ -528,26 +539,25 @@ srs_error_t SrsRtcServer::create_session(
     return err;
 }
 
-srs_error_t SrsRtcServer::do_create_session(
-    SrsRtcConnection* session, SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp, const std::string& mock_eip,
-    bool publish, bool dtls, bool srtp
-)
+srs_error_t SrsRtcServer::do_create_session(SrsRtcUserConfig* ruc, SrsSdp& local_sdp, SrsRtcConnection* session)
 {
     srs_error_t err = srs_success;
 
+    SrsRequest* req = ruc->req_;
+
     // first add publisher/player for negotiate sdp media info
-    if (publish) {
-        if ((err = session->add_publisher(req, remote_sdp, local_sdp)) != srs_success) {
+    if (ruc->publish_) {
+        if ((err = session->add_publisher(ruc, local_sdp)) != srs_success) {
             return srs_error_wrap(err, "add publisher");
         }
     } else {
-        if ((err = session->add_player(req, remote_sdp, local_sdp)) != srs_success) {
+        if ((err = session->add_player(ruc, local_sdp)) != srs_success) {
             return srs_error_wrap(err, "add player");
         }
     }
 
     // All tracks default as inactive, so we must enable them.
-    session->set_all_tracks_status(req->get_stream_url(), publish, true);
+    session->set_all_tracks_status(req->get_stream_url(), ruc->publish_, true);
 
     std::string local_pwd = srs_random_str(32);
     std::string local_ufrag = "";
@@ -556,7 +566,7 @@ srs_error_t SrsRtcServer::do_create_session(
     while (true) {
         local_ufrag = srs_random_str(8);
 
-        username = local_ufrag + ":" + remote_sdp.get_ice_ufrag();
+        username = local_ufrag + ":" + ruc->remote_sdp_.get_ice_ufrag();
         if (!_srs_rtc_manager->find_by_name(username)) {
             break;
         }
@@ -568,13 +578,13 @@ srs_error_t SrsRtcServer::do_create_session(
     local_sdp.set_fingerprint(_srs_rtc_dtls_certificate->get_fingerprint());
 
     // We allows to mock the eip of server.
-    if (!mock_eip.empty()) {
+    if (!ruc->eip_.empty()) {
         string host;
         int port = _srs_config->get_rtc_server_listen();
-        srs_parse_hostport(mock_eip, host, port);
+        srs_parse_hostport(ruc->eip_, host, port);
 
         local_sdp.add_candidate(host, port, "host");
-        srs_trace("RTC: Use candidate mock_eip %s as %s:%d", mock_eip.c_str(), host.c_str(), port);
+        srs_trace("RTC: Use candidate mock_eip %s as %s:%d", ruc->eip_.c_str(), host.c_str(), port);
     } else {
         std::vector<string> candidate_ips = get_candidate_ips();
         for (int i = 0; i < (int)candidate_ips.size(); ++i) {
@@ -587,11 +597,11 @@ srs_error_t SrsRtcServer::do_create_session(
     local_sdp.session_negotiate_ = local_sdp.session_config_;
 
     // Setup the negotiate DTLS role.
-    if (remote_sdp.get_dtls_role() == "active") {
+    if (ruc->remote_sdp_.get_dtls_role() == "active") {
         local_sdp.session_negotiate_.dtls_role = "passive";
-    } else if (remote_sdp.get_dtls_role() == "passive") {
+    } else if (ruc->remote_sdp_.get_dtls_role() == "passive") {
         local_sdp.session_negotiate_.dtls_role = "active";
-    } else if (remote_sdp.get_dtls_role() == "actpass") {
+    } else if (ruc->remote_sdp_.get_dtls_role() == "actpass") {
         local_sdp.session_negotiate_.dtls_role = local_sdp.session_config_.dtls_role;
     } else {
         // @see: https://tools.ietf.org/html/rfc4145#section-4.1
@@ -601,13 +611,13 @@ srs_error_t SrsRtcServer::do_create_session(
     }
     local_sdp.set_dtls_role(local_sdp.session_negotiate_.dtls_role);
 
-    session->set_remote_sdp(remote_sdp);
+    session->set_remote_sdp(ruc->remote_sdp_);
     // We must setup the local SDP, then initialize the session object.
     session->set_local_sdp(local_sdp);
     session->set_state(WAITING_STUN);
 
     // Before session initialize, we must setup the local SDP.
-    if ((err = session->initialize(req, dtls, srtp, username)) != srs_success) {
+    if ((err = session->initialize(req, ruc->dtls_, ruc->srtp_, username)) != srs_success) {
         return srs_error_wrap(err, "init");
     }
 
diff --git a/trunk/src/app/srs_app_rtc_server.hpp b/trunk/src/app/srs_app_rtc_server.hpp
index 075b9d1ed..ef3064dbb 100644
--- a/trunk/src/app/srs_app_rtc_server.hpp
+++ b/trunk/src/app/srs_app_rtc_server.hpp
@@ -31,6 +31,7 @@
 #include <srs_app_reload.hpp>
 #include <srs_app_hourglass.hpp>
 #include <srs_app_hybrid.hpp>
+#include <srs_app_rtc_sdp.hpp>
 
 #include <string>
 
@@ -84,6 +85,25 @@ public:
     virtual srs_error_t on_udp_packet(SrsUdpMuxSocket* skt, SrsRtcConnection* session, bool* pconsumed) = 0;
 };
 
+// The user config for RTC publish or play.
+class SrsRtcUserConfig
+{
+public:
+    // Original variables from API.
+    SrsSdp remote_sdp_;
+    std::string eip_;
+    std::string codec_;
+
+    // Generated data.
+    SrsRequest* req_;
+    bool publish_;
+    bool dtls_;
+    bool srtp_;
+public:
+    SrsRtcUserConfig();
+    virtual ~SrsRtcUserConfig();
+};
+
 // The RTC server instance, listen UDP port, handle UDP packet, manage RTC connections.
 class SrsRtcServer : public ISrsUdpMuxHandler, public ISrsFastTimer, public ISrsReloadHandler
 {
@@ -111,16 +131,9 @@ public:
     srs_error_t listen_api();
 public:
     // Peer start offering, we answer it.
-    srs_error_t create_session(
-        SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp, const std::string& mock_eip,
-        bool publish, bool dtls, bool srtp,
-        SrsRtcConnection** psession
-    );
+    srs_error_t create_session(SrsRtcUserConfig* ruc, SrsSdp& local_sdp, SrsRtcConnection** psession);
 private:
-    srs_error_t do_create_session(
-        SrsRtcConnection* session, SrsRequest* req, const SrsSdp& remote_sdp, SrsSdp& local_sdp,
-        const std::string& mock_eip, bool publish, bool dtls, bool srtp
-    );
+    srs_error_t do_create_session(SrsRtcUserConfig* ruc, SrsSdp& local_sdp, SrsRtcConnection* session);
 public:
     SrsRtcConnection* find_session_by_username(const std::string& ufrag);
 // interface ISrsFastTimer
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index 8a2e0bb42..441c0fa7d 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    90
+#define VERSION_REVISION    91
 
 #endif