diff --git a/projects/IoTClient/Assets/StreamingAssets/wwwroot/device.html b/projects/IoTClient/Assets/StreamingAssets/wwwroot/device.html
index 9fa74035..8868dd37 100644
--- a/projects/IoTClient/Assets/StreamingAssets/wwwroot/device.html
+++ b/projects/IoTClient/Assets/StreamingAssets/wwwroot/device.html
@@ -20,7 +20,7 @@
                 </f7-col>
                 <f7-page>
                     <f7-navbar :title="model?model.Name:'节点'">
-                        <a v-if="model" :href="'product.html?number='+model.Product.Number" slot="nav-left" class="link external"><i class="icon icon-back"></i></a>
+                        <a v-if="model" :href="'product.html?number='+model.Product.Number" slot="nav-left" class="link external"><i class="icon icon-back"></i>返回</a>
                     </f7-navbar>
                     <div id="template">
                         <style>
diff --git a/projects/IoTClient/Assets/StreamingAssets/wwwroot/js/page.js b/projects/IoTClient/Assets/StreamingAssets/wwwroot/js/page.js
index c35560f2..060d6e50 100644
--- a/projects/IoTClient/Assets/StreamingAssets/wwwroot/js/page.js
+++ b/projects/IoTClient/Assets/StreamingAssets/wwwroot/js/page.js
@@ -1,9 +1,9 @@
 //
-var version = '1.0.0-rc.106';
+var version = '1.0.0-beta.1227';
 var isDebug = isDebug || false;
 var useSignalR = useSignalR || false;
 var userAgent = navigator.userAgent;
-var isApp = userAgent.indexOf('isapp') !== 0;
+var isApp = userAgent.indexOf('isapp') !== -1||location.href.indexOf('file://') !== -1;
 var isAndroid = userAgent.indexOf('isandroid')!==-1;
 var isIos=userAgent.indexOf('isiphone')!==-1;
 //
@@ -27,131 +27,131 @@ var muted = true;
 var volume = 0.5;
 //
 if (isApp) {
-    Framework7.use(Framework7Vue);
+	Framework7.use(Framework7Vue);
 }
 toastr.options.timeOut = 200;
 toastr.options.positionClass = "toast-top-center";
 var f7params = isApp? {
-    routes: [],
-    name: 'My App',
-    id: 'com.myapp.test',
-    theme: 'ios',
+	routes: [],
+	name: 'My App',
+	id: 'com.myapp.test',
+	theme: 'ios',
 }:null;
 //
 //log
 function debug(message) {
-    if (isDebug) {
-        console.log(message);
-    }
+	if (isDebug) {
+		console.log(message);
+	}
 }
 //signalr
 function connect() {
-    debug('start connect to server:' + Date());
-    if (connection.state === signalR.HubConnectionState.Disconnected) {
-        connection.start().then(function () {
-            toastr.success('客户端与服务器连接成功！');
-            load();
-        }).catch(function (err) {
-            toastr.error('客户端与服务器连接失败！');
-            location.href = "config.html";
-            setTimeout(connect, 15 * 1000);
-        });
-    }
+	debug('start connect to server:' + Date());
+	if (connection.state === signalR.HubConnectionState.Disconnected) {
+		connection.start().then(function () {
+			toastr.success('客户端与服务器连接成功！');
+			load();
+		}).catch(function (err) {
+			toastr.error('客户端与服务器连接失败！');
+			location.href = "config.html";
+			setTimeout(connect, 15 * 1000);
+		});
+	}
 }
 function signalRInit() {
-    connection.on('Connected', function (id) {
-        connectionId = id;
-        debug(connectionId);
-    });
-    connection.onclose(function (err) {
-        setTimeout(connect, 15 * 1000);
-    });
-    connection.on("ServerToClient", function (method, json, to,from) {
-        debug(method + ':' + json);
-        onMessage(method, json,to, from);
-    });
+	connection.on('Connected', function (id) {
+		connectionId = id;
+		debug(connectionId);
+	});
+	connection.onclose(function (err) {
+		setTimeout(connect, 15 * 1000);
+	});
+	connection.on("ServerToClient", function (method, json, to,from) {
+		debug(method + ':' + json);
+		onMessage(method, json,to, from);
+	});
 }
 //加载模板
 function loadTemplate() {
-    var formData = new FormData();
-    if (isApp) {
-        formData.append('token', token);
-    }
-    axios.post(templateUrl,formData)
-        .then(function (response) {
-            var html = response.data;
-            if (html) {
-                $('#template').html(html);
-            }
-            init();
-        })
-        .catch(function (error) {
-            toastr.error('模板加载失败');
-            debug(error);
-        });
+	var formData = new FormData();
+	if (isApp) {
+		formData.append('token', token);
+	}
+	axios.post(templateUrl,formData)
+		.then(function (response) {
+			var html = response.data;
+			if (html) {
+				$('#template').html(html);
+			}
+			init();
+		})
+		.catch(function (error) {
+			toastr.error('模板加载失败');
+			debug(error);
+		});
 }
 //加载数据
 function loadData() {
-    var formData = new FormData();
-    if (isApp) {
-        formData.append('token', token);
-    }
-    axios.post(dataUrl, formData)
-        .then(function (response) {
-            debug(response);
-            vm.model = response.data;
-            toastr.success('数据加载成功！');
-            loadDataSuccess();
-        })
-        .catch(function (error) {
-            toastr.error('数据加载失败');
-            debug(error);
-        });
+	var formData = new FormData();
+	if (isApp) {
+		formData.append('token', token);
+	}
+	axios.post(dataUrl, formData)
+		.then(function (response) {
+			debug(response);
+			vm.model = response.data;
+			toastr.success('数据加载成功！');
+			loadDataSuccess();
+		})
+		.catch(function (error) {
+			toastr.error('数据加载失败');
+			debug(error);
+		});
 }
 //更新数据
 function updateItem(list, item) {
-    var update = false;
-    for (var i = 0; i < list.length; i++) {
-        if (list[i].Id == item.Id) {
-            update = true;
-            break;
-        }
-    }
-    if (update) {
-        list.splice(i, 1, item);
-    }
-    else {
-        list.push(item);
-    }
+	var update = false;
+	for (var i = 0; i < list.length; i++) {
+		if (list[i].Id == item.Id) {
+			update = true;
+			break;
+		}
+	}
+	if (update) {
+		list.splice(i, 1, item);
+	}
+	else {
+		list.push(item);
+	}
 }
 function deleteItem(list, item) {
-    for (var i = 0; i < list.length; i++) {
-        if (list[i].Id == item.Id) {
-            list.splice(i, 1);
-            break;
-        }
-    }
+	for (var i = 0; i < list.length; i++) {
+		if (list[i].Id == item.Id) {
+			list.splice(i, 1);
+			break;
+		}
+	}
 }
 //
 //开始
 function start() {
-    if (useSignalR) {
-        connection = new signalR.HubConnectionBuilder().withUrl(wsUrl).build();
-        signalRInit();
-        connect();
-    }
-    else {
-        load();
-    }
+	if (useSignalR) {
+		connection = new signalR.HubConnectionBuilder().withUrl(wsUrl).build();
+		signalRInit();
+		connect();
+	}
+	else {
+		load();
+	}
 }
 //
 function load() {
-    if (templateUrl) {
-        loadTemplate();
-    }
-    else {
-        init();
-    }
+	if (templateUrl) {
+		loadTemplate();
+	}
+	else {
+		init();
+	}
 }
 //初始化vue
 var data = function () { };
@@ -159,487 +159,487 @@ var mounted = function () { };
 var loadDataSuccess = function () { };
 var methods ={ };
 function init() {
-    $('#template style').each(function (i, n) {
-        $('head').append(n);
-    });
-    vm = new Vue({
-        el: '#app',
-        data: data,
-        mounted: function () {
-            $('#loading').remove();
-            if (isApp) {
-                if (!this.server && location.href.indexOf('config.html') === -1) {
-                    location.href = "config.html";
-                }
-                if (!this.token) {
-                    if (location.href.indexOf('config.html') === -1 && location.href.indexOf('login.html') === -1) {
-                        location.href = "login.html";
-                    }
-                }
-                mounted();
-                this.$f7ready((f7) => {
-                    app = this.$f7;
-                    $.validator.unobtrusive.parse('form');
-                });
-            }
-            if (dataUrl) {
-                loadData();
-            }
-        },
-        methods: methods
-    });
+	$('#template style').each(function (i, n) {
+		$('head').append(n);
+	});
+	vm = new Vue({
+		el: '#app',
+		data: data,
+		mounted: function () {
+			$('#loading').remove();
+			if (isApp) {
+				if (!this.server && location.href.indexOf('config.html') === -1) {
+					location.href = "config.html";
+				}
+				if (!this.token) {
+					if (location.href.indexOf('config.html') === -1 && location.href.indexOf('login.html') === -1) {
+						location.href = "login.html";
+					}
+				}
+				mounted();
+				this.$f7ready((f7) => {
+					app = this.$f7;
+					$.validator.unobtrusive.parse('form');
+				});
+			}
+			if (dataUrl) {
+				loadData();
+			}
+		},
+		methods: methods
+	});
 }
 //tools
 //
 function getCode1(code) {
-    var pattern = 0;
-    var direction = 0;
-    var wind = 0;
-    var power = 2;
-    var temperature = 1;
-    if (code > 480 + 2) {
-        pattern = 480;
-    }
-    else if (code > 360 + 2) {
-        pattern = 360;
-    }
-    else if (code > 240 + 2) {
-        pattern = 240;
-    }
-    else if (code > 120 + 2) {
-        pattern = 120;
-    }
-    else {
-        pattern = 0;
-    }
-    code -= pattern;
-    if (code > 75 + 2) {
-        direction = 75;
-    }
-    else if (code > 60 + 2) {
-        direction = 60;
-    }
-    else {
-        direction = 0;
-    }
-    code -= direction;
-    if (code > 45 + 2) {
-        wind = 45;
-    }
-    else if (code > 30 + 2) {
-        wind = 30;
-    }
-    else if (code > 15 + 2) {
-        wind = 15;
-    }
-    else {
-        wind = 0;
-    }
-    code -= wind;
-    if (code === 1) {
-        power = 1;
-    }
-    else {
-        power = 2;
-        temperature = code - power;
-    }
-    return {
-        pattern: pattern,
-        direction: direction,
-        wind: wind,
-        power: power,
-        temperature: temperature
-    };
+	var pattern = 0;
+	var direction = 0;
+	var wind = 0;
+	var power = 2;
+	var temperature = 1;
+	if (code > 480 + 2) {
+		pattern = 480;
+	}
+	else if (code > 360 + 2) {
+		pattern = 360;
+	}
+	else if (code > 240 + 2) {
+		pattern = 240;
+	}
+	else if (code > 120 + 2) {
+		pattern = 120;
+	}
+	else {
+		pattern = 0;
+	}
+	code -= pattern;
+	if (code > 75 + 2) {
+		direction = 75;
+	}
+	else if (code > 60 + 2) {
+		direction = 60;
+	}
+	else {
+		direction = 0;
+	}
+	code -= direction;
+	if (code > 45 + 2) {
+		wind = 45;
+	}
+	else if (code > 30 + 2) {
+		wind = 30;
+	}
+	else if (code > 15 + 2) {
+		wind = 15;
+	}
+	else {
+		wind = 0;
+	}
+	code -= wind;
+	if (code === 1) {
+		power = 1;
+	}
+	else {
+		power = 2;
+		temperature = code - power;
+	}
+	return {
+		pattern: pattern,
+		direction: direction,
+		wind: wind,
+		power: power,
+		temperature: temperature
+	};
 }
 //
 function hsvToRgb(h, s, v) {
-    if (h == "") h = 0;
-    if (s == "") s = 0;
-    if (v == "") v = 0;
-    h = parseFloat(h);
-    s = parseFloat(s);
-    v = parseFloat(v);
-    if (h < 0) h = 0;
-    if (s < 0) s = 0;
-    if (v < 0) v = 0;
-    if (h >= 360) h = 359;
-    if (s > 100) s = 100;
-    if (v > 100) v = 100;
-    s /= 100.0;
-    v /= 100.0;
-    C = v * s;
-    hh = h / 60.0;
-    X = C * (1.0 - Math.abs((hh % 2) - 1.0));
-    r = g = b = 0;
-    if (hh >= 0 && hh < 1) {
-        r = C;
-        g = X;
-    }
-    else if (hh >= 1 && hh < 2) {
-        r = X;
-        g = C;
-    }
-    else if (hh >= 2 && hh < 3) {
-        g = C;
-        b = X;
-    }
-    else if (hh >= 3 && hh < 4) {
-        g = X;
-        b = C;
-    }
-    else if (hh >= 4 && hh < 5) {
-        r = X;
-        b = C;
-    }
-    else {
-        r = C;
-        b = X;
-    }
-    m = v - C;
-    r += m;
-    g += m;
-    b += m;
-    r *= 255.0;
-    g *= 255.0;
-    b *= 255.0;
-    r = Math.round(r);
-    g = Math.round(g);
-    b = Math.round(b);
-    hex = r * 65536 + g * 256 + b;
-    hex = hex.toString(16, 6);
-    len = hex.length;
-    if (len < 6)
-        for (i = 0; i < 6 - len; i++)
-            hex = '0' + hex;
-    return '#' + hex;
+	if (h == "") h = 0;
+	if (s == "") s = 0;
+	if (v == "") v = 0;
+	h = parseFloat(h);
+	s = parseFloat(s);
+	v = parseFloat(v);
+	if (h < 0) h = 0;
+	if (s < 0) s = 0;
+	if (v < 0) v = 0;
+	if (h >= 360) h = 359;
+	if (s > 100) s = 100;
+	if (v > 100) v = 100;
+	s /= 100.0;
+	v /= 100.0;
+	C = v * s;
+	hh = h / 60.0;
+	X = C * (1.0 - Math.abs((hh % 2) - 1.0));
+	r = g = b = 0;
+	if (hh >= 0 && hh < 1) {
+		r = C;
+		g = X;
+	}
+	else if (hh >= 1 && hh < 2) {
+		r = X;
+		g = C;
+	}
+	else if (hh >= 2 && hh < 3) {
+		g = C;
+		b = X;
+	}
+	else if (hh >= 3 && hh < 4) {
+		g = X;
+		b = C;
+	}
+	else if (hh >= 4 && hh < 5) {
+		r = X;
+		b = C;
+	}
+	else {
+		r = C;
+		b = X;
+	}
+	m = v - C;
+	r += m;
+	g += m;
+	b += m;
+	r *= 255.0;
+	g *= 255.0;
+	b *= 255.0;
+	r = Math.round(r);
+	g = Math.round(g);
+	b = Math.round(b);
+	hex = r * 65536 + g * 256 + b;
+	hex = hex.toString(16, 6);
+	len = hex.length;
+	if (len < 6)
+		for (i = 0; i < 6 - len; i++)
+			hex = '0' + hex;
+	return '#' + hex;
 }
 //
 function ajax(url, data, type) {
-    url = iotCenter + url
-    console.log(url);
-    type = type || 'get';
-    $('.overlay').show();
-    $.ajax({
-        type: type,
-        url: url,
-        data: data,
-        success: AjaxCallBack
-    }).fail(function (result) {
-        toastr.error('请求发送失败！');
-        console.log(result);
-    }).always(function () {
-        $('.overlay').hide();
-    });
+	url = iotCenter + url
+	console.log(url);
+	type = type || 'get';
+	$('.overlay').show();
+	$.ajax({
+		type: type,
+		url: url,
+		data: data,
+		success: AjaxCallBack
+	}).fail(function (result) {
+		toastr.error('请求发送失败！');
+		console.log(result);
+	}).always(function () {
+		$('.overlay').hide();
+	});
 }
 //
 function AjaxCallBack(response) {
-    var result = response;
-    if (result.code === 0) {
-        if (result.type === 0) {
-            if (result.format === 1) {
-                console.log('format/1/base64 jpeg image');
-                $('#callback .page-content').html('<img class="shot" src="' + result.data + '">');
-            }
-            else {
-                console.log('format/0/json object');
-                $('#callback .page-content').html(result.data);
-            }
-            app.popup.open('#callback');
-        }
-    }
+	var result = response;
+	if (result.code === 0) {
+		if (result.type === 0) {
+			if (result.format === 1) {
+				console.log('format/1/base64 jpeg image');
+				$('#callback .page-content').html('<img class="shot" src="' + result.data + '">');
+			}
+			else {
+				console.log('format/0/json object');
+				$('#callback .page-content').html(result.data);
+			}
+			app.popup.open('#callback');
+		}
+	}
 }
 //
 function hsvToRgb(h, s, v) {
-    if (h == "") h = 0;
-    if (s == "") s = 0;
-    if (v == "") v = 0;
-    h = parseFloat(h);
-    s = parseFloat(s);
-    v = parseFloat(v);
-    if (h < 0) h = 0;
-    if (s < 0) s = 0;
-    if (v < 0) v = 0;
-    if (h >= 360) h = 359;
-    if (s > 100) s = 100;
-    if (v > 100) v = 100;
-    s /= 100.0;
-    v /= 100.0;
-    C = v * s;
-    hh = h / 60.0;
-    X = C * (1.0 - Math.abs((hh % 2) - 1.0));
-    r = g = b = 0;
-    if (hh >= 0 && hh < 1) {
-        r = C;
-        g = X;
-    }
-    else if (hh >= 1 && hh < 2) {
-        r = X;
-        g = C;
-    }
-    else if (hh >= 2 && hh < 3) {
-        g = C;
-        b = X;
-    }
-    else if (hh >= 3 && hh < 4) {
-        g = X;
-        b = C;
-    }
-    else if (hh >= 4 && hh < 5) {
-        r = X;
-        b = C;
-    }
-    else {
-        r = C;
-        b = X;
-    }
-    m = v - C;
-    r += m;
-    g += m;
-    b += m;
-    r *= 255.0;
-    g *= 255.0;
-    b *= 255.0;
-    r = Math.round(r);
-    g = Math.round(g);
-    b = Math.round(b);
-    hex = r * 65536 + g * 256 + b;
-    hex = hex.toString(16, 6);
-    len = hex.length;
-    if (len < 6)
-        for (i = 0; i < 6 - len; i++)
-            hex = '0' + hex;
-    return '#' + hex;
+	if (h == "") h = 0;
+	if (s == "") s = 0;
+	if (v == "") v = 0;
+	h = parseFloat(h);
+	s = parseFloat(s);
+	v = parseFloat(v);
+	if (h < 0) h = 0;
+	if (s < 0) s = 0;
+	if (v < 0) v = 0;
+	if (h >= 360) h = 359;
+	if (s > 100) s = 100;
+	if (v > 100) v = 100;
+	s /= 100.0;
+	v /= 100.0;
+	C = v * s;
+	hh = h / 60.0;
+	X = C * (1.0 - Math.abs((hh % 2) - 1.0));
+	r = g = b = 0;
+	if (hh >= 0 && hh < 1) {
+		r = C;
+		g = X;
+	}
+	else if (hh >= 1 && hh < 2) {
+		r = X;
+		g = C;
+	}
+	else if (hh >= 2 && hh < 3) {
+		g = C;
+		b = X;
+	}
+	else if (hh >= 3 && hh < 4) {
+		g = X;
+		b = C;
+	}
+	else if (hh >= 4 && hh < 5) {
+		r = X;
+		b = C;
+	}
+	else {
+		r = C;
+		b = X;
+	}
+	m = v - C;
+	r += m;
+	g += m;
+	b += m;
+	r *= 255.0;
+	g *= 255.0;
+	b *= 255.0;
+	r = Math.round(r);
+	g = Math.round(g);
+	b = Math.round(b);
+	hex = r * 65536 + g * 256 + b;
+	hex = hex.toString(16, 6);
+	len = hex.length;
+	if (len < 6)
+		for (i = 0; i < 6 - len; i++)
+			hex = '0' + hex;
+	return '#' + hex;
 }
 //
 function changeTime(t) {
-    time = t;
-    UpdateChart();
+	time = t;
+	UpdateChart();
 }
 function UpdateChart(deviceNumber) {
-    $('canvas.chart').each(function () {
-        var canvas = this;
-        var number = canvas.id;
-        if (!deviceNumber || deviceNumber === number) {
-            var device = vm.GetDevice(number);
-            var url = iotCenter + '/App/GetChartData?time=' + time + '&number=' + number + "&token=" + token + "&connectionId=" + connectionId;
-            axios.post(url, { crossDomain: true })
-                .then(function (response) {
-                    var data = response.data;
-                    UpdateChartInternal(canvas, device.DisplayName, data);
-                    console.log('折线图更新成功');
-                })
-                .catch(function (error) {
-                    console.log(error);
-                    toastr.error(error)
-                });
-        }
-    });
+	$('canvas.chart').each(function () {
+		var canvas = this;
+		var number = canvas.id;
+		if (!deviceNumber || deviceNumber === number) {
+			var device = vm.GetDevice(number);
+			var url = iotCenter + '/App/GetChartData?time=' + time + '&number=' + number + "&token=" + token + "&connectionId=" + connectionId;
+			axios.post(url, { crossDomain: true })
+				.then(function (response) {
+					var data = response.data;
+					UpdateChartInternal(canvas, device.DisplayName, data);
+					console.log('折线图更新成功');
+				})
+				.catch(function (error) {
+					console.log(error);
+					toastr.error(error)
+				});
+		}
+	});
 }
 function UpdateChartInternal(canvas, title, data) {
-    var chart;
-    Chart.helpers.each(Chart.instances, function (instance) {
-        if (instance.chart.canvas.id === canvas.id) {
-            chart = instance;
-        }
-    })
-    if (chart) {
-        chart.data = data;
-        chart.update();
-    }
-    else {
-        var ctx = canvas.getContext('2d');
-        var options = {
-            responsive: true,
-            legend: {
-                position: 'bottom',
-            },
-            title: {
-                display: false,
-                text: title
-            },
-            animation: {
-                duration: 0
-            }
-        };
-        var chart = new Chart(ctx, {
-            type: 'line',
-            data: data,
-            options: options
-        });
-    }
+	var chart;
+	Chart.helpers.each(Chart.instances, function (instance) {
+		if (instance.chart.canvas.id === canvas.id) {
+			chart = instance;
+		}
+	})
+	if (chart) {
+		chart.data = data;
+		chart.update();
+	}
+	else {
+		var ctx = canvas.getContext('2d');
+		var options = {
+			responsive: true,
+			legend: {
+				position: 'bottom',
+			},
+			title: {
+				display: false,
+				text: title
+			},
+			animation: {
+				duration: 0
+			}
+		};
+		var chart = new Chart(ctx, {
+			type: 'line',
+			data: data,
+			options: options
+		});
+	}
 }
 //
 function UpdateCamera() {
-    var number = $('#camera').val();
-    var video = document.getElementById("liveVideo");
-    if (number && video) {
-        if (flvjs.isSupported) {
+	var number = $('#camera').val();
+	var video = document.getElementById("liveVideo");
+	if (number && video) {
+		if (flvjs.isSupported) {
 			if(isApp&&!isAndroid)
 			{
-				playFlv(video);
+				playHls(video);
 			}
 			else{
-				playHls(video);
+				playFlv(video); 
 			}
-        }
-        else {
-            playHls(video);
-        }
-    }
+		}
+		else {
+			playHls(video);
+		}
+	}
 }
-function playHls(videoElement) {
-    var number = $('#camera').val();
-    var device = vm.model.Devices
-        ? Enumerable.from(vm.model.Devices).where(function (o) { return o.Number === number; }).firstOrDefault()
-        : vm.model;
-    if (device) {
-        var name = isApp ? "子码流hls" : "主码流hls";
-        var data = Enumerable.from(device.Data).where(o => o.Name === name).firstOrDefault();
-        if (data) {
-            var url = data.Value;
-            var hasPtz = Enumerable.from(device.Data).where(o => o.Name === '缩放支持').firstOrDefault();
-            if (hasPtz && hasPtz.Value === '是') {
-                $('#ptz').show();
-            }
-            else {
-                $('#ptz').hide();
-            }
-            video.src = url;
-        }
-    }
+function playHls(video) {
+	var number = $('#camera').val();
+	var device = vm.model.Devices
+		? Enumerable.from(vm.model.Devices).where(function (o) { return o.Number === number; }).firstOrDefault()
+		: vm.model;
+	if (device) {
+		var name = isApp ? "子码流hls" : "主码流hls";
+		var data = Enumerable.from(device.Data).where(o => o.Name === name).firstOrDefault();
+		if (data) {
+			var url = data.Value;
+			var hasPtz = Enumerable.from(device.Data).where(o => o.Name === '缩放支持').firstOrDefault();
+			if (hasPtz && hasPtz.Value === '是') {
+				$('#ptz').show();
+			}
+			else {
+				$('#ptz').hide();
+			}
+			video.src = url;
+		}
+	}
 }
 function playFlv(videoElement) {
-    var number = $('#camera').val();
-    if (number) {
-        var device = vm.model.Devices
-            ? Enumerable.from(vm.model.Devices).where(function (o) { return o.Number === number; }).firstOrDefault()
-            : vm.model;
-        var name = isApp ? "子码流flv" : "主码流flv";
-        var data = Enumerable.from(device.Data).where(o => o.Name === name).firstOrDefault();
-        if (data) {
-            var url = data.Value;
-            var hasPtz = Enumerable.from(device.Data).where(o => o.Name === '缩放支持').firstOrDefault();
-            if (hasPtz && hasPtz.Value === '是') {
-                $('#ptz').show();
-            }
-            else {
-                $('#ptz').hide();
-            }
-            closePlayer(livePlayer);
-            var push = Enumerable.from(device.Data).where(o => o.Name === '推流').firstOrDefault();
-            if (push && push.Value === '是') {
-                try {
-                    livePlayer = flvjs.createPlayer({
-                        type: 'flv',
-                        url: url,
-                        isLive: true,
-                        cors: true
-                    }, {
-                        enableWorker: true,
-                        enableStashBuffer: false,
-                        stashInitialSize: 1,
-                        fixAudioTimestampGap: false
-                    });
-                    livePlayer.attachMediaElement(videoElement);
-                    livePlayer.load();
-                    livePlayer.volume = volume;
-                    livePlayer.muted = muted;
-                    timer = setInterval(function () {
-                        console.log('.');
-                        if (livePlayer.statisticsInfo.speed == 0) {
-                            console.log('reload1');
-                            clearInterval(timer);
-                            playFlv(videoElement);
-                        }
-                        else if (decodedFrames && livePlayer.statisticsInfo.decodedFrames <= decodedFrames) {
-                            console.log('reload2');
-                            clearInterval(timer);
-                            playFlv(videoElement);
-                        }
-                        else if (livePlayer.buffered.end(0) - livePlayer.currentTime > 1) {
-                            console.log('reset currentTime');
-                            livePlayer.currentTime = livePlayer.buffered.end(0) - 0.001;
-                        }
-                        decodedFrames = livePlayer.statisticsInfo.decodedFrames;
-                    }, 10 * 1000);
-                } catch (e) {
-                    console.log(e);
-                }
-            }
-        }
-    }
+	var number = $('#camera').val();
+	if (number) {
+		var device = vm.model.Devices
+			? Enumerable.from(vm.model.Devices).where(function (o) { return o.Number === number; }).firstOrDefault()
+			: vm.model;
+		var name = isApp ? "子码流flv" : "主码流flv";
+		var data = Enumerable.from(device.Data).where(o => o.Name === name).firstOrDefault();
+		if (data) {
+			var url = data.Value;
+			var hasPtz = Enumerable.from(device.Data).where(o => o.Name === '缩放支持').firstOrDefault();
+			if (hasPtz && hasPtz.Value === '是') {
+				$('#ptz').show();
+			}
+			else {
+				$('#ptz').hide();
+			}
+			closePlayer(livePlayer);
+			var push = Enumerable.from(device.Data).where(o => o.Name === '推流').firstOrDefault();
+			if (push && push.Value === '是') {
+				try {
+					livePlayer = flvjs.createPlayer({
+						type: 'flv',
+						url: url,
+						isLive: true,
+						cors: true
+					}, {
+						enableWorker: true,
+						enableStashBuffer: false,
+						stashInitialSize: 1,
+						fixAudioTimestampGap: false
+					});
+					livePlayer.attachMediaElement(videoElement);
+					livePlayer.load();
+					livePlayer.volume = volume;
+					livePlayer.muted = muted;
+					timer = setInterval(function () {
+						console.log('.');
+						if (livePlayer.statisticsInfo.speed == 0) {
+							console.log('reload1');
+							clearInterval(timer);
+							playFlv(videoElement);
+						}
+						else if (decodedFrames && livePlayer.statisticsInfo.decodedFrames <= decodedFrames) {
+							console.log('reload2');
+							clearInterval(timer);
+							playFlv(videoElement);
+						}
+						else if (livePlayer.buffered.end(0) - livePlayer.currentTime > 1) {
+							console.log('reset currentTime');
+							livePlayer.currentTime = livePlayer.buffered.end(0) - 0.001;
+						}
+						decodedFrames = livePlayer.statisticsInfo.decodedFrames;
+					}, 10 * 1000);
+				} catch (e) {
+					console.log(e);
+				}
+			}
+		}
+	}
 }
 function closePlayer(livePlayer) {
-    if (timer) {
-        clearInterval(timer);
-    }
-    if (livePlayer != null) {
-        try {
-            livePlayer.pause();
-        } catch (e) {
-            console.error(e);
-        }
-        livePlayer.unload();
-        livePlayer.detachMediaElement();
-        try {
-            livePlayer.destroy();
-        } catch (e) {
-            console.error(e);
-        }
-        livePlayer = null;
-    }
+	if (timer) {
+		clearInterval(timer);
+	}
+	if (livePlayer != null) {
+		try {
+			livePlayer.pause();
+		} catch (e) {
+			console.error(e);
+		}
+		livePlayer.unload();
+		livePlayer.detachMediaElement();
+		try {
+			livePlayer.destroy();
+		} catch (e) {
+			console.error(e);
+		}
+		livePlayer = null;
+	}
 }
 $(document).on('change', '#camera', function (e) {
-    UpdateCamera()
+	UpdateCamera()
 });
 $(document).on('mousedown touchstart', 'img.camera.ajax', function (e) {
-    var number = $('#camera').val();
-    var cmd = $(this).attr('data-cmd');
-    ajax('/App/ExecApi', { token: token, connectionId: connectionId, number: number, method: '/Onvif/' + cmd }, 'post');
-    return false;
+	var number = $('#camera').val();
+	var cmd = $(this).attr('data-cmd');
+	ajax('/App/ExecApi', { token: token, connectionId: connectionId, number: number, method: '/Onvif/' + cmd }, 'post');
+	return false;
 });
 $(document).on('mouseup touchend', 'img.camera.ajax', function (e) {
-    var number = $('#camera').val();
-    var cmd = $(this).attr('data-cmd');
-    if (cmd !== 'Stop') {
-        setTimeout(function () {
-            ajax('/App/ExecApi', { token: token, connectionId: connectionId, number: number, method: '/Onvif/Stop' }, 'post');
-        }, 500);
-    }
-    return false;
+	var number = $('#camera').val();
+	var cmd = $(this).attr('data-cmd');
+	if (cmd !== 'Stop') {
+		setTimeout(function () {
+			ajax('/App/ExecApi', { token: token, connectionId: connectionId, number: number, method: '/Onvif/Stop' }, 'post');
+		}, 500);
+	}
+	return false;
 });
 $(document).on('click', '.segmented .button', function (e) {
-    var current = $(this);
-    if (!current.hasClass('button-active')) {
-        current.parent().find('.button-active').removeClass('button-active');
-        current.addClass('button-active');
-    }
+	var current = $(this);
+	if (!current.hasClass('button-active')) {
+		current.parent().find('.button-active').removeClass('button-active');
+		current.addClass('button-active');
+	}
 });
 function Select(e) {
-    var checkbox = $(e.target);
-    if (checkbox.hasClass('checkall')) {
-        if (e.target.checked) {
-            $('input.item:visible').not(':checked').prop("checked", true);
-        }
-        else {
-            $('input.item:visible').filter(':checked').prop("checked", false);
-        }
-    }
-    else if (checkbox.hasClass('uncheck')) {
-        $('input.item:visible').each(function () {
-            $(this).prop("checked", !$(this).prop("checked")).change();
-        });
-    }
-    var parent = $('input.checkall');
-    if ($('input.item').not(':checked').length === 0) {
-        parent.prop("indeterminate", false);
-        parent.prop("checked", true);
-    }
-    else if ($('input.item').filter(':checked').length === 0) {
-        parent.prop("indeterminate", false);
-        parent.prop("checked", false);
-    }
-    else {
-        parent.prop("indeterminate", true);
-    }
+	var checkbox = $(e.target);
+	if (checkbox.hasClass('checkall')) {
+		if (e.target.checked) {
+			$('input.item:visible').not(':checked').prop("checked", true);
+		}
+		else {
+			$('input.item:visible').filter(':checked').prop("checked", false);
+		}
+	}
+	else if (checkbox.hasClass('uncheck')) {
+		$('input.item:visible').each(function () {
+			$(this).prop("checked", !$(this).prop("checked")).change();
+		});
+	}
+	var parent = $('input.checkall');
+	if ($('input.item').not(':checked').length === 0) {
+		parent.prop("indeterminate", false);
+		parent.prop("checked", true);
+	}
+	else if ($('input.item').filter(':checked').length === 0) {
+		parent.prop("indeterminate", false);
+		parent.prop("checked", false);
+	}
+	else {
+		parent.prop("indeterminate", true);
+	}
 }
diff --git a/projects/IoTClient/Assets/StreamingAssets/wwwroot/node.html b/projects/IoTClient/Assets/StreamingAssets/wwwroot/node.html
index 2e06dbab..63cedec8 100644
--- a/projects/IoTClient/Assets/StreamingAssets/wwwroot/node.html
+++ b/projects/IoTClient/Assets/StreamingAssets/wwwroot/node.html
@@ -20,7 +20,7 @@
                 </f7-col>
                 <f7-page>
                     <f7-navbar :title="model?model.Name:'节点'">
-                        <a href="nodes.html" slot="nav-left" class="link external"><i class="icon icon-back"></i></a>
+                        <a href="nodes.html" slot="nav-left" class="link external"><i class="icon icon-back"></i>返回</a>
                     </f7-navbar>
                     <div id="template">
                         <style>
diff --git a/projects/IoTClient/Assets/StreamingAssets/wwwroot/product.html b/projects/IoTClient/Assets/StreamingAssets/wwwroot/product.html
index a92e1606..df297147 100644
--- a/projects/IoTClient/Assets/StreamingAssets/wwwroot/product.html
+++ b/projects/IoTClient/Assets/StreamingAssets/wwwroot/product.html
@@ -17,7 +17,7 @@
             <f7-view main>
                 <f7-page>
                     <f7-navbar :title="model?model.Name:'产品'">
-                        <a href="index.html" slot="nav-left" class="link external"><i class="icon icon-back"></i></a>
+                        <a href="index.html" slot="nav-left" class="link external"><i class="icon icon-back"></i>返回</a>
                     </f7-navbar>
                     <div class="row" v-if="model">
                         <div class="col-100" v-if="HasBatchCommand()">
diff --git a/projects/IoTClient/Assets/StreamingAssets/wwwroot/user.html b/projects/IoTClient/Assets/StreamingAssets/wwwroot/user.html
index a45742c3..25a72251 100644
--- a/projects/IoTClient/Assets/StreamingAssets/wwwroot/user.html
+++ b/projects/IoTClient/Assets/StreamingAssets/wwwroot/user.html
@@ -38,7 +38,7 @@
                             <button class="col button" type="submit" v-on:click="logout">退出</button>
                         </div>
                     </div>
-                    <div>v {{version}}</div>
+                    <div class="block block-strong" style="text-align:center;">version {{version}}</div>
                     <f7-toolbar tabbar labels position="bottom">
                         <a href="index.html" class="tab-link external">
                             <i class="icon icon f7-icons">house</i>
diff --git a/projects/IoTClient/ProjectSettings/GraphicsSettings.asset b/projects/IoTClient/ProjectSettings/GraphicsSettings.asset
index 349be978..491588c1 100644
--- a/projects/IoTClient/ProjectSettings/GraphicsSettings.asset
+++ b/projects/IoTClient/ProjectSettings/GraphicsSettings.asset
@@ -39,6 +39,7 @@ GraphicsSettings:
   - {fileID: 16000, guid: 0000000000000000f000000000000000, type: 0}
   - {fileID: 17000, guid: 0000000000000000f000000000000000, type: 0}
   - {fileID: 16001, guid: 0000000000000000f000000000000000, type: 0}
+  - {fileID: 16003, guid: 0000000000000000f000000000000000, type: 0}
   m_PreloadedShaders: []
   m_SpritesDefaultMaterial: {fileID: 10754, guid: 0000000000000000f000000000000000,
     type: 0}
diff --git a/projects/IoTClient/iotclient.apk b/projects/IoTClient/iotclient.apk
index 66367de0..01e57aad 100644
Binary files a/projects/IoTClient/iotclient.apk and b/projects/IoTClient/iotclient.apk differ
diff --git a/projects/IoTNode/DeviceServices/FBee/ClusterId.cs b/projects/IoTNode/DeviceServices/FBee/ClusterId.cs
index b8a56b7a..26559440 100644
--- a/projects/IoTNode/DeviceServices/FBee/ClusterId.cs
+++ b/projects/IoTNode/DeviceServices/FBee/ClusterId.cs
@@ -10,6 +10,7 @@
         humidity = 0x0405,
         voltage = 0x0001,
         socket = 0x0702,
-        SummationDivisor = 0x0b04
+        SummationDivisor = 0x0b04,
+        doorlock = 0x0101
     }
 }
\ No newline at end of file
diff --git a/projects/IoTNode/DeviceServices/FBee/DeviceId.cs b/projects/IoTNode/DeviceServices/FBee/DeviceId.cs
index 85d5e41d..c5e76c80 100644
--- a/projects/IoTNode/DeviceServices/FBee/DeviceId.cs
+++ b/projects/IoTNode/DeviceServices/FBee/DeviceId.cs
@@ -10,6 +10,7 @@ namespace IoTNode.DeviceServices.FBee
         {
             List.Add(new DeviceId { RawDeviceId = 0x0002, Name = "开关", CategoryName = "照明", CategoryNumber = "30", Icon = "switch" });
             List.Add(new DeviceId { RawDeviceId = 0x0009, Name = "插座", CategoryName = "电器", CategoryNumber = "20", Icon = "socket" });
+            List.Add(new DeviceId { RawDeviceId = 0x000a, Name = "门锁", CategoryName = "安防", CategoryNumber = "10", Icon = "lock" });
             List.Add(new DeviceId { RawDeviceId = 0x0051, Name = "智能插座", CategoryName = "电器", CategoryNumber = "20", Icon = "socket" });
             List.Add(new DeviceId { RawDeviceId = 0x0106, Name = "光强检测器", CategoryName = "监测", CategoryNumber = "40", Icon = "light" });
             List.Add(new DeviceId { RawDeviceId = 0x0163, Name = "红外转发器", CategoryName = "电器", CategoryNumber = "20", Icon = "control" });
diff --git a/projects/IoTNode/DeviceServices/FBee/FBeeService.cs b/projects/IoTNode/DeviceServices/FBee/FBeeService.cs
index c33f72b1..ad030843 100644
--- a/projects/IoTNode/DeviceServices/FBee/FBeeService.cs
+++ b/projects/IoTNode/DeviceServices/FBee/FBeeService.cs
@@ -596,6 +596,9 @@ namespace IoTNode.DeviceServices.FBee
                             //deviceRepo.SaveChanges();
                             //var deviceDto = device.To<EditDeviceModel>();
                             //this.SendToServer(Methods.EditDevice, deviceDto);
+                            if (deviceId == 0x00a)
+                            {
+                            }
                             this.UpdateStatus(device);
                         }
                         else
@@ -645,6 +648,10 @@ namespace IoTNode.DeviceServices.FBee
             {
                 return "/Ir/";
             }
+            else if (deviceName == "门锁")
+            {
+                return "/Lock/";
+            }
             else
             {
                 return null;
@@ -1310,6 +1317,9 @@ namespace IoTNode.DeviceServices.FBee
                                 this.UpdateData(deviceRepo, device, device.CreateData(Keys.Electricity, electricity, DeviceDataType.Float, "电量", "kWh", timestamp: timestamp));
                             }
                         }
+                        else if (clusterId == ClusterId.doorlock)
+                        {
+                        }
                         if (deviceId == 0x0163)
                         {
                             if (props.ContainsKey(0x400a))
diff --git a/projects/gateway/src/main/resources/application.yml b/projects/gateway/src/main/resources/application.yml
index 36d2ba4c..0ca298ed 100644
--- a/projects/gateway/src/main/resources/application.yml
+++ b/projects/gateway/src/main/resources/application.yml
@@ -13,13 +13,20 @@ logging:
 spring:
   cloud:
     gateway:
-      #globalcors:
-      #corsConfigurations:
-      #'[/**]':
-      #allowedOrigins: "*"
-      #allowedHeaders: "*"
-      #allowedMethods: "*"
       routes:
+        - id: SRS
+          uri: http://localhost:8080/
+          predicates:
+            - Path=/live/**
+          filters:
+            - StripPrefix=1
+            - PrefixPath=/live
+        - id: InfluxDB
+          uri: http://localhost:8086/
+          predicates:
+            - Path=/influxdb/**
+          filters:
+            - StripPrefix=1
         - id: UserCenter
           uri: http://localhost:8010/
           predicates:
diff --git a/publish/src/linux-arm64/publish/apps/srs/conf/full.conf b/publish/src/linux-arm64/publish/apps/srs/conf/full.conf
deleted file mode 100644
index cf3e0217..00000000
--- a/publish/src/linux-arm64/publish/apps/srs/conf/full.conf
+++ /dev/null
@@ -1,1848 +0,0 @@
-# all config for srs
-
-#############################################################################################
-# RTMP sections
-#############################################################################################
-# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
-# for example, 192.168.1.100:1935 10.10.10.100:1935
-# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
-listen              1935;
-# the pid file
-# to ensure only one process can use a pid file
-# and provides the current running process id, for script,
-# for example, init.d script to manage the server.
-# default: ./objs/srs.pid
-pid                 ./objs/srs.pid;
-# the default chunk size is 128, max is 65536,
-# some client does not support chunk size change,
-# however, most clients support it and it can improve
-# performance about 10%.
-# default: 60000
-chunk_size          60000;
-# the logs dir.
-# if enabled ffmpeg, each transcoding stream will create a log file.
-# /dev/null to disable the log.
-# default: ./objs
-ff_log_dir          ./objs;
-# the log tank, console or file.
-# if console, print log to console.
-# if file, write log to file. requires srs_log_file if log to file.
-# default: file.
-srs_log_tank        file;
-# the log level, for all log tanks.
-# can be: verbose, info, trace, warn, error
-# default: trace
-srs_log_level       trace;
-# when srs_log_tank is file, specifies the log file.
-# default: ./objs/srs.log
-srs_log_file        ./objs/srs.log;
-# the max connections.
-# if exceed the max connections, server will drop the new connection.
-# default: 1000
-max_connections     1000;
-# whether start as daemon
-# @remark: do not support reload.
-# default: on
-daemon              on;
-# whether use utc_time to generate the time struct,
-# if off, use localtime() to generate it,
-# if on, use gmtime() instead, which use UTC time.
-# default: off
-utc_time            off;
-# config for the pithy print in ms,
-# which always print constant message specified by interval,
-# whatever the clients in concurrency.
-# default: 10000
-pithy_print_ms      10000;
-
-# the work dir for server, to chdir(work_dir) when not empty or "./"
-# user can config this directory to change the dir.
-# @reamrk do not support reload.
-# default: ./
-work_dir ./;
-# whether quit when parent process changed,
-# used for supervisor mode(not daemon), srs should always quit when 
-# supervisor process exited.
-# @remark conflict with daemon, error when both daemon and asprocess are on.
-# @reamrk do not support reload.
-# default: off
-asprocess off;
-
-#############################################################################################
-# heartbeat/stats sections
-#############################################################################################
-# heartbeat to api server
-# @remark, the ip report to server, is retrieve from system stat,
-#       which need the config item stats.network.
-heartbeat {
-    # whether heartbeat is enabled.
-    # default: off
-    enabled         off;
-    # the interval seconds for heartbeat,
-    # recommend 0.3,0.6,0.9,1.2,1.5,1.8,2.1,2.4,2.7,3,...,6,9,12,....
-    # default: 9.9
-    interval        9.3;
-    # when startup, srs will heartbeat to this api.
-    # @remark: must be a restful http api url, where SRS will POST with following data:
-    #   {
-    #       "device_id": "my-srs-device",
-    #       "ip": "192.168.1.100"
-    #   }
-    # default: http://127.0.0.1:8085/api/v1/servers
-    url             http://127.0.0.1:8085/api/v1/servers;
-    # the id of device.
-    device_id       "my-srs-device";
-    # whether report with summaries
-    # if on, put /api/v1/summaries to the request data:
-    #   {
-    #       "summaries": summaries object.
-    #   }
-    # @remark: optional config.
-    # default: off
-    summaries       off;
-}
-
-# system statistics section.
-# the main cycle will retrieve the system stat,
-# for example, the cpu/mem/network/disk-io data,
-# the http api, for instance, /api/v1/summaries will show these data.
-# @remark the heartbeat depends on the network,
-#       for example, the eth0 maybe the device which index is 0.
-stats {
-    # the index of device ip.
-    # we may retrieve more than one network device.
-    # default: 0
-    network         0;
-    # the device name to stat the disk iops.
-    # ignore the device of /proc/diskstats if not configured.
-    disk            sda sdb xvda xvdb;
-}
-
-#############################################################################################
-# HTTP sections
-#############################################################################################
-# api of srs.
-# the http api config, export for external program to manage srs.
-# user can access http api of srs in browser directly, for instance, to access by:
-#       curl http://192.168.1.170:1985/api/v1/reload
-# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
-# where the cli can only be used in shell/terminate.
-http_api {
-    # whether http api is enabled.
-    # default: off
-    enabled         on;
-    # the http api listen entry is <[ip:]port>
-    # for example, 192.168.1.100:1985
-    # where the ip is optional, default to 0.0.0.0, that is 1985 equals to 0.0.0.0:1985
-    # default: 1985
-    listen          1985;
-    # whether enable crossdomain request.
-    # default: on
-    crossdomain     on;
-    # the HTTP RAW API is more powerful api to change srs state and reload.
-    raw_api {
-        # whether enable the HTTP RAW API.
-        # default: off
-        enabled             off;
-        # whether enable rpc reload.
-        # default: off
-        allow_reload        off;
-        # whether enable rpc query.
-        # default: off
-        allow_query         off;
-        # whether enable rpc update.
-        # default: off
-        allow_update        off;
-    }
-}
-# embedded http server in srs.
-# the http streaming config, for HLS/HDS/DASH/HTTPProgressive
-# global config for http streaming, user must config the http section for each vhost.
-# the embed http server used to substitute nginx in ./objs/nginx,
-# for example, srs running in arm, can provides RTMP and HTTP service, only with srs installed.
-# user can access the http server pages, generally:
-#       curl http://192.168.1.170:80/srs.html
-# which will show srs version and welcome to srs.
-# @remark, the http embedded stream need to config the vhost, for instance, the __defaultVhost__
-# need to open the feature http of vhost.
-http_server {
-    # whether http streaming service is enabled.
-    # default: off
-    enabled         on;
-    # the http streaming listen entry is <[ip:]port>
-    # for example, 192.168.1.100:8080
-    # where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
-    # @remark, if use lower port, for instance 80, user must start srs by root.
-    # default: 8080
-    listen          8080;
-    # the default dir for http root.
-    # default: ./objs/nginx/html
-    dir             ./objs/nginx/html;
-    # whether enable crossdomain request.
-    # for both http static and stream server and apply on all vhosts.
-    # default: on
-    crossdomain     on;
-}
-
-#############################################################################################
-# Streamer sections
-#############################################################################################
-# the streamer cast stream from other protocol to SRS over RTMP.
-# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
-stream_caster {
-    # whether stream caster is enabled.
-    # default: off
-    enabled         off;
-    # the caster type of stream, the casters:
-    #       mpegts_over_udp, MPEG-TS over UDP caster.
-    #       rtsp, Real Time Streaming Protocol (RTSP).
-    #       flv, FLV over HTTP POST.
-    caster          mpegts_over_udp;
-    # the output rtmp url.
-    # for mpegts_over_udp caster, the typically output url:
-    #           rtmp://127.0.0.1/live/livestream
-    # for rtsp caster, the typically output url:
-    #           rtmp://127.0.0.1/[app]/[stream]
-    #       for example, the rtsp url:
-    #           rtsp://192.168.1.173:8544/live/livestream.sdp
-    #       where the [app] is "live" and [stream] is "livestream", output is:
-    #           rtmp://127.0.0.1/live/livestream
-    # for flv caster, the typically output url:
-    #           rtmp://127.0.0.1/[app]/[stream]
-    #       for example, POST to url:
-    #           http://127.0.0.1:8936/live/livestream.flv
-    #       where the [app] is "live" and [stream] is "livestream", output is:
-    #           rtmp://127.0.0.1/live/livestream
-    output          rtmp://127.0.0.1/live/livestream;
-    # the listen port for stream caster.
-    #       for mpegts_over_udp caster, listen at udp port. for example, 8935.
-    #       for rtsp caster, listen at tcp port. for example, 554.
-    #       for flv caster, listen at tcp port. for example, 8936.
-    # TODO: support listen at <[ip:]port>
-    listen          8935;
-    # for the rtsp caster, the rtp server local port over udp,
-    # which reply the rtsp setup request message, the port will be used:
-    #       [rtp_port_min, rtp_port_max)
-    rtp_port_min    57200;
-    rtp_port_max    57300;
-}
-stream_caster {
-    enabled         off;
-    caster          mpegts_over_udp;
-    output          rtmp://127.0.0.1/live/livestream;
-    listen          8935;
-}
-stream_caster {
-    enabled         off;
-    caster          rtsp;
-    output          rtmp://127.0.0.1/[app]/[stream];
-    listen          554;
-    rtp_port_min    57200;
-    rtp_port_max    57300;
-}
-stream_caster {
-    enabled         off;
-    caster          flv;
-    output          rtmp://127.0.0.1/[app]/[stream];
-    listen          8936;
-}
-
-#############################################################################################
-# Kafka sections
-#############################################################################################
-# Apache Kafka is a high-throughput distributed messaging system.
-# SRS is a Kafka producer to send message to kafka.
-# @see https://kafka.apache.org/documentation.html#introduction
-kafka {
-    # whether enabled kafka.
-    # default: off
-    enabled         off;
-    # the broker list, broker is <ip:port>
-    # and use space to specify multple brokers.
-    # for example, 127.0.0.1:9092 127.0.0.1:9093
-    brokers         127.0.0.1:9092;
-    # the kafka topic to use.
-    # default: srs
-    topic           srs;
-}
-
-#############################################################################################
-# RTMP/HTTP VHOST sections
-#############################################################################################
-# vhost list, the __defaultVhost__ is the default vhost
-# for example, user use ip to access the stream: rtmp://192.168.1.2/live/livestream.
-# for which cannot identify the required vhost.
-vhost __defaultVhost__ {
-}
-
-# the vhost scope configs.
-vhost scope.vhost.srs.com {
-    # whether the vhost is enabled.
-    # if off, all request access denied.
-    # default: on
-    enabled         off;
-
-    # whether enable min delay mode for vhost.
-    # for min latency mode:
-    # 1. disable the publish.mr for vhost.
-    # 2. use timeout for cond wait for consumer queue.
-    # @see https://github.com/ossrs/srs/issues/257
-    # default: off
-    min_latency     off;
-
-    # whether enable the TCP_NODELAY
-    # if on, set the nodelay of fd by setsockopt
-    # default: off
-    tcp_nodelay     off;
-
-    # the default chunk size is 128, max is 65536,
-    # some client does not support chunk size change,
-    # vhost chunk size will override the global value.
-    # default: global chunk size.
-    chunk_size      128;
-    
-    # The input ack size, 0 to not set.
-    # Generally, it's set by the message from peer,
-    # but for some peer(encoder), it never send message but use a different ack size.
-    # We can chnage the default ack size in server-side, to send acknowledge message,
-    # or the encoder maybe blocked after publishing for some time.
-    # Default: 0
-    in_ack_size     0;
-    
-    # The output ack size, 0 to not set.
-    # This is used to notify the peer(player) to send acknowledge to server.
-    # Default: 2500000
-    out_ack_size    2500000;
-}
-
-# set the chunk size of vhost.
-vhost chunksize.srs.com {
-    # @see scope.vhost.srs.com
-    chunk_size      128;
-}
-
-# the vhost disabled.
-vhost removed.srs.com {
-    # @see scope.vhost.srs.com
-    enabled         off;
-}
-
-# vhost for stream cluster for RTMP/FLV
-vhost cluster.srs.com {
-    # The config for cluster.
-    cluster {
-        # The cluster mode, local or remote.
-        #       local: It's an origin server, serve streams itself.
-        #       remote: It's an edge server, fetch or push stream to origin server.
-        # default: local
-        mode            remote;
-
-        # For edge(mode remote), user must specifies the origin server
-        # format as: <server_name|ip>[:port]
-        # @remark user can specifies multiple origin for error backup, by space,
-        # for example, 192.168.1.100:1935 192.168.1.101:1935 192.168.1.102:1935
-        origin          127.0.0.1:1935 localhost:1935;
-
-        # For edge(mode remote), whether open the token traverse mode,
-        # if token traverse on, all connections of edge will forward to origin to check(auth),
-        # it's very important for the edge to do the token auth.
-        # the better way is use http callback to do the token auth by the edge,
-        # but if user prefer origin check(auth), the token_traverse if better solution.
-        # default: off
-        token_traverse  off;
-
-        # For edge(mode remote), the vhost to transform for edge,
-        # to fetch from the specified vhost at origin,
-        # if not specified, use the current vhost of edge in origin, the variable [vhost].
-        # default: [vhost]
-        vhost           same.edge.srs.com;
-
-        # For edge(mode remote), when upnode(forward to, edge push to, edge pull from) is srs,
-        # it's strongly recommend to open the debug_srs_upnode,
-        # when connect to upnode, it will take the debug info,
-        # for example, the id, source id, pid.
-        # please see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLog
-        # default: on
-        debug_srs_upnode    on;
-
-        # For origin(mode local) cluster, turn on the cluster.
-        # @remark Origin cluster only supports RTMP, use Edge to transmux RTMP to FLV.
-        # default: off
-        # TODO: FIXME: Support reload.
-        origin_cluster      off;
-
-        # For origin (mode local) cluster, the co-worker's HTTP APIs.
-        # This origin will connect to co-workers and communicate with them.
-        # please read: https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
-        # TODO: FIXME: Support reload.
-        coworkers           127.0.0.1:9091 127.0.0.1:9092;
-    }
-}
-
-# vhost for edge, edge and origin is the same vhost
-vhost same.edge.srs.com {
-    # @see cluster.srs.com
-    cluster {
-        mode            remote;
-        origin          127.0.0.1:1935 localhost:1935;
-        token_traverse  off;
-    }
-}
-
-# vhost for edge, edge transform vhost to fetch from another vhost.
-vhost transform.edge.srs.com {
-    # @see cluster.srs.com
-    cluster {
-        mode            remote;
-        origin          127.0.0.1:1935;
-        vhost           same.edge.srs.com;
-    }
-}
-
-# the vhost for srs debug info, whether send args in connect(tcUrl).
-vhost debug.srs.com {
-    # @see cluster.srs.com
-    cluster {
-        debug_srs_upnode    on;
-    }
-}
-
-# the vhost which forward publish streams.
-vhost same.vhost.forward.srs.com {
-    # forward stream to other servers.
-    forward {
-        # whether enable the forward.
-        # default: off
-        enabled on;
-        # forward all publish stream to the specified server.
-        # this used to split/forward the current stream for cluster active-standby,
-        # active-active for cdn to build high available fault tolerance system.
-        # format: {ip}:{port} {ip_N}:{port_N}
-        destination 127.0.0.1:1936 127.0.0.1:1937;
-    }
-}
-
-# the play specified configs
-vhost play.srs.com {
-    # for play client, both RTMP and other stream clients,
-    # for instance, the HTTP FLV stream clients.
-    play {
-        # whether cache the last gop.
-        # if on, cache the last gop and dispatch to client,
-        #   to enabled fast startup for client, client play immediately.
-        # if off, send the latest media data to client,
-        #   client need to wait for the next Iframe to decode and show the video.
-        # set to off if requires min delay;
-        # set to on if requires client fast startup.
-        # default: on
-        gop_cache       off;
-        # the max live queue length in seconds.
-        # if the messages in the queue exceed the max length,
-        # drop the old whole gop.
-        # default: 30
-        queue_length    10;
-
-        # about the stream monotonically increasing:
-        #   1. video timestamp is monotonically increasing,
-        #   2. audio timestamp is monotonically increasing,
-        #   3. video and audio timestamp is interleaved/mixed monotonically increasing.
-        # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
-        # however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
-        # the time jitter algorithm:
-        #   1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
-        #   2. zero, only ensure stream start at zero, ignore timestamp jitter.
-        #   3. off, disable the time jitter algorithm, like atc.
-        # @remark for full, correct timestamp only when |delta| > 250ms.
-        # @remark disabled when atc is on.
-        # default: full
-        time_jitter             full;
-        # vhost for atc for hls/hds/rtmp backup.
-        # generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0.
-        # when atc is on, server delivery rtmp stream by absolute time.
-        # atc is used, for instance, encoder will copy stream to master and slave server,
-        # server use atc to delivery stream to edge/client, where stream time from master/slave server
-        # is always the same, client/tools can slice RTMP stream to HLS according to the same time,
-        # if the time not the same, the HLS stream cannot slice to support system backup.
-        #
-        # @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html
-        # @see http://www.baidu.com/#wd=hds%20hls%20atc
-        #
-        # @remark when atc is on, auto off the time_jitter
-        # default: off
-        atc             off;
-        # whether use the interleaved/mixed algorithm to correct the timestamp.
-        # if on, always ensure the timestamp of audio+video is interleaved/mixed monotonically increase.
-        # if off, use time_jitter to correct the timestamp if required.
-        # @remark to use mix_correct, atc should on(or time_jitter should off).
-        # default: off
-        mix_correct             off;
-
-        # whether enable the auto atc,
-        # if enabled, detect the bravo_atc="true" in onMetaData packet,
-        # set atc to on if matched.
-        # always ignore the onMetaData if atc_auto is off.
-        # default: off
-        atc_auto        off;
-
-        # set the MW(merged-write) latency in ms.
-        # SRS always set mw on, so we just set the latency value.
-        # the latency of stream >= mw_latency + mr_latency
-        # the value recomment is [300, 1800]
-        # default: 350
-        mw_latency      350;
-
-        # the minimal packets send interval in ms,
-        # used to control the ndiff of stream by srs_rtmp_dump,
-        # for example, some device can only accept some stream which
-        # delivery packets in constant interval(not cbr).
-        # @remark 0 to disable the minimal interval.
-        # @remark >0 to make the srs to send message one by one.
-        # @remark user can get the right packets interval in ms by srs_rtmp_dump.
-        # default: 0
-        send_min_interval       10.0;
-        # whether reduce the sequence header,
-        # for some client which cannot got duplicated sequence header,
-        # while the sequence header is not changed yet.
-        # default: off
-        reduce_sequence_header  on;
-    }
-}
-
-# vhost for time jitter
-vhost jitter.srs.com {
-    # @see play.srs.com
-    # to use time_jitter full, the default config.
-    play {
-    }
-    # to use mix_correct.
-    play {
-        time_jitter             off;
-        mix_correct             on;
-    }
-    play {
-        atc                     on;
-        mix_correct             on;
-    }
-    # to use atc
-    play {
-        atc                     on;
-    }
-}
-
-# vhost for atc.
-vhost atc.srs.com {
-    # @see play.srs.com
-    play {
-        atc             on;
-        atc_auto        on;
-    }
-}
-
-# the MR(merged-read) setting for publisher.
-# the MW(merged-write) settings for player.
-vhost mrw.srs.com {
-    # @see scope.vhost.srs.com
-    min_latency     off;
-
-    # @see play.srs.com
-    play {
-        mw_latency      350;
-    }
-
-    # @see publish.srs.com
-    publish {
-        mr          on;
-        mr_latency  350;
-    }
-}
-
-# the vhost for min delay, do not cache any stream.
-vhost min.delay.com {
-    # @see scope.vhost.srs.com
-    min_latency     on;
-    # @see scope.vhost.srs.com
-    tcp_nodelay     on;
-
-    # @see play.srs.com
-    play {
-        mw_latency      100;
-        gop_cache       off;
-        queue_length    10;
-    }
-
-    # @see publish.srs.com
-    publish {
-        mr          off;
-    }
-}
-
-# whether disable the sps parse, for the resolution of video.
-vhost no.parse.sps.com {
-    # @see publish.srs.com
-    publish {
-        parse_sps   on;
-    }
-}
-
-# the vhost to control the stream delivery feature
-vhost stream.control.com {
-    # @see scope.vhost.srs.com
-    min_latency     on;
-    # @see scope.vhost.srs.com
-    tcp_nodelay     on;
-
-    # @see play.srs.com
-    play {
-        mw_latency      100;
-        queue_length    10;
-        send_min_interval       10.0;
-        reduce_sequence_header  on;
-    }
-
-    # @see publish.srs.com
-    publish {
-        mr          off;
-        firstpkt_timeout    20000;
-        normal_timeout      7000;
-    }
-}
-
-# the publish specified configs
-vhost publish.srs.com {
-    # the config for FMLE/Flash publisher, which push RTMP to SRS.
-    publish {
-        # about MR, read https://github.com/ossrs/srs/issues/241
-        # when enabled the mr, SRS will read as large as possible.
-        # default: off
-        mr          off;
-        # the latency in ms for MR(merged-read),
-        # the performance+ when latency+, and memory+,
-        #       memory(buffer) = latency * kbps / 8
-        # for example, latency=500ms, kbps=3000kbps, each publish connection will consume
-        #       memory = 500 * 3000 / 8 = 187500B = 183KB
-        # when there are 2500 publisher, the total memory of SRS at least:
-        #       183KB * 2500 = 446MB
-        # the recommended value is [300, 2000]
-        # default: 350
-        mr_latency  350;
-
-        # the 1st packet timeout in ms for encoder.
-        # default: 20000
-        firstpkt_timeout    20000;
-        # the normal packet timeout in ms for encoder.
-        # default: 5000
-        normal_timeout      7000;
-        # whether parse the sps when publish stream.
-        # we can got the resolution of video for stat api.
-        # but we may failed to cause publish failed.
-        # default: on
-        parse_sps   on;
-    }
-}
-
-# the vhost for anti-suck.
-vhost refer.anti_suck.com {
-    # refer hotlink-denial.
-    refer {
-        # whether enable the refer hotlink-denial.
-        # default: off.
-        enabled         on;
-        # the common refer for play and publish.
-        # if the page url of client not in the refer, access denied.
-        # if not specified this field, allow all.
-        # default: not specified.
-        all           github.com github.io;
-        # refer for publish clients specified.
-        # the common refer is not overridden by this.
-        # if not specified this field, allow all.
-        # default: not specified.
-        publish   github.com github.io;
-        # refer for play clients specified.
-        # the common refer is not overridden by this.
-        # if not specified this field, allow all.
-        # default: not specified.
-        play      github.com github.io;
-    }
-}
-
-# vhost for bandwidth check
-# generally, the bandcheck vhost must be: bandcheck.srs.com,
-# or need to modify the vhost of client.
-vhost bandcheck.srs.com {
-    enabled         on;
-    chunk_size      65000;
-    # bandwidth check config.
-    bandcheck {
-        # whether support bandwidth check,
-        # default: off.
-        enabled         on;
-        # the key for server to valid,
-        # if invalid key, server disconnect and abort the bandwidth check.
-        key             "35c9b402c12a7246868752e2878f7e0e";
-        # the interval in seconds for bandwidth check,
-        # server do not allow new test request.
-        # default: 30
-        interval        30;
-        # the max available check bandwidth in kbps.
-        # to avoid attack of bandwidth check.
-        # default: 1000
-        limit_kbps      4000;
-    }
-}
-
-# the security to allow or deny clients.
-vhost security.srs.com {
-    # security for host to allow or deny clients.
-    # @see https://github.com/ossrs/srs/issues/211
-    security {
-        # whether enable the security for vhost.
-        # default: off
-        enabled         on;
-        # the security list, each item format as:
-        #       allow|deny    publish|play    all|<ip>
-        # for example:
-        #       allow           publish     all;
-        #       deny            publish     all;
-        #       allow           publish     127.0.0.1;
-        #       deny            publish     127.0.0.1;
-        #       allow           play        all;
-        #       deny            play        all;
-        #       allow           play        127.0.0.1;
-        #       deny            play        127.0.0.1;
-        # SRS apply the following simple strategies one by one:
-        #       1. allow all if security disabled.
-        #       2. default to deny all when security enabled.
-        #       3. allow if matches allow strategy.
-        #       4. deny if matches deny strategy.
-        allow           play        all;
-        allow           publish     all;
-    }
-}
-
-# vhost for http static and flv vod stream for each vhost.
-vhost http.static.srs.com {
-    # http static vhost specified config
-    http_static {
-        # whether enabled the http static service for vhost.
-        # default: off
-        enabled     on;
-        # the url to mount to,
-        # typical mount to [vhost]/
-        # the variables:
-        #       [vhost] current vhost for http server.
-        # @remark the [vhost] is optional, used to mount at specified vhost.
-        # @remark the http of __defaultVhost__ will override the http_server section.
-        # for example:
-        #       mount to [vhost]/
-        #           access by http://ossrs.net:8080/xxx.html
-        #       mount to [vhost]/hls
-        #           access by http://ossrs.net:8080/hls/xxx.html
-        #       mount to /
-        #           access by http://ossrs.net:8080/xxx.html
-        #           or by http://192.168.1.173:8080/xxx.html
-        #       mount to /hls
-        #           access by http://ossrs.net:8080/hls/xxx.html
-        #           or by http://192.168.1.173:8080/hls/xxx.html
-        # @remark the port of http is specified by http_server section.
-        # default: [vhost]/
-        mount       [vhost]/hls;
-        # main dir of vhost,
-        # to delivery HTTP stream of this vhost.
-        # default: ./objs/nginx/html
-        dir         ./objs/nginx/html/hls;
-    }
-}
-
-# vhost for http flv/aac/mp3 live stream for each vhost.
-vhost http.remux.srs.com {
-    # http flv/mp3/aac/ts stream vhost specified config
-    http_remux {
-        # whether enable the http live streaming service for vhost.
-        # default: off
-        enabled     on;
-        # the fast cache for audio stream(mp3/aac),
-        # to cache more audio and send to client in a time to make android(weixin) happy.
-        # @remark the flv/ts stream ignore it
-        # @remark 0 to disable fast cache for http audio stream.
-        # default: 0
-        fast_cache  30;
-        # the stream mount for rtmp to remux to live streaming.
-        # typical mount to [vhost]/[app]/[stream].flv
-        # the variables:
-        #       [vhost] current vhost for http live stream.
-        #       [app] current app for http live stream.
-        #       [stream] current stream for http live stream.
-        # @remark the [vhost] is optional, used to mount at specified vhost.
-        # the extension:
-        #       .flv mount http live flv stream, use default gop cache.
-        #       .ts mount http live ts stream, use default gop cache.
-        #       .mp3 mount http live mp3 stream, ignore video and audio mp3 codec required.
-        #       .aac mount http live aac stream, ignore video and audio aac codec required.
-        # for example:
-        #       mount to [vhost]/[app]/[stream].flv
-        #           access by http://ossrs.net:8080/live/livestream.flv
-        #       mount to /[app]/[stream].flv
-        #           access by http://ossrs.net:8080/live/livestream.flv
-        #           or by http://192.168.1.173:8080/live/livestream.flv
-        #       mount to [vhost]/[app]/[stream].mp3
-        #           access by http://ossrs.net:8080/live/livestream.mp3
-        #       mount to [vhost]/[app]/[stream].aac
-        #           access by http://ossrs.net:8080/live/livestream.aac
-        #       mount to [vhost]/[app]/[stream].ts
-        #           access by http://ossrs.net:8080/live/livestream.ts
-        # @remark the port of http is specified by http_server section.
-        # default: [vhost]/[app]/[stream].flv
-        mount       [vhost]/[app]/[stream].flv;
-    }
-}
-
-# the http hook callback vhost, srs will invoke the hooks for specified events.
-vhost hooks.callback.srs.com {
-    http_hooks {
-        # whether the http hooks enable.
-        # default off.
-        enabled         on;
-        # when client connect to vhost/app, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_connect",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
-        #           "pageUrl": "http://www.test.com/live.html"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_connect      http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
-        # when client close/disconnect to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_close",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "send_bytes": 10240, "recv_bytes": 10240
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_close        http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
-        # when client(encoder) publish to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_publish",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_publish http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_publish      http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
-        # when client(encoder) stop publish to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_unpublish",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_unpublish http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_unpublish    http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
-        # when client start to play vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_play",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy",
-        #           "pageUrl": "http://www.test.com/live.html"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_play http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_play         http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
-        # when client stop to play vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_stop",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_stop http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_stop         http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
-        # when srs reap a dvr file, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_dvr",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy",
-        #           "cwd": "/usr/local/srs",
-        #           "file": "./objs/nginx/html/live/livestream.1420254068776.flv"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        on_dvr          http://127.0.0.1:8085/api/v1/dvrs http://localhost:8085/api/v1/dvrs;
-        # when srs reap a ts file of hls, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_hls",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy",
-        #           "duration": 9.36, // in seconds
-        #           "cwd": "/usr/local/srs",
-        #           "file": "./objs/nginx/html/live/livestream/2015-04-23/01/476584165.ts",
-        #           "url": "live/livestream/2015-04-23/01/476584165.ts",
-        #           "m3u8": "./objs/nginx/html/live/livestream/live.m3u8",
-        #           "m3u8_url": "live/livestream/live.m3u8",
-        #           "seq_no": 100
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        on_hls          http://127.0.0.1:8085/api/v1/hls http://localhost:8085/api/v1/hls;
-        # when srs reap a ts file of hls, call this hook,
-        # used to push file to cdn network, by get the ts file from cdn network.
-        # so we use HTTP GET and use the variable following:
-        #       [app], replace with the app.
-        #       [stream], replace with the stream.
-        #       [param], replace with the param.
-        #       [ts_url], replace with the ts url.
-        # ignore any return data of server.
-        # @remark random select a url to report, not report all.
-        on_hls_notify   http://127.0.0.1:8085/api/v1/hls/[app]/[stream]/[ts_url][param];
-    }
-}
-
-# the vhost for exec, fork process when publish stream.
-vhost exec.srs.com {
-    # the exec used to fork process when got some event.
-    exec {
-        # whether enable the exec.
-        # default: off.
-        enabled     off;
-        # when publish stream, exec the process with variables:
-        #       [vhost] the input stream vhost.
-        #       [port] the input stream port.
-        #       [app] the input stream app.
-        #       [stream] the input stream name.
-        #       [engine] the transcode engine name.
-        # other variables for exec only:
-        #       [url] the rtmp url which trigger the publish.
-        #       [tcUrl] the client request tcUrl.
-        #       [swfUrl] the client request swfUrl.
-        #       [pageUrl] the client request pageUrl.
-        # @remark empty to ignore this exec.
-        publish     ./objs/ffmpeg/bin/ffmpeg -f flv -i [url] -c copy -y ./[stream].flv;
-    }
-}
-
-# The vhost for MPEG-DASH.
-vhost dash.srs.com {
-    dash {
-        # Whether DASH is enabled.
-        # Transmux RTMP to DASH if on.
-        # Default: off
-        enabled             on;
-        # The duration of segment in seconds.
-        # Default: 3
-        dash_fragment       3;
-        # The period to update the MPD in seconds.
-        # Default: 30
-        dash_update_period  30;
-        # The depth of timeshift buffer in seconds.
-        # Default: 60
-        dash_timeshift      60;
-        # The base/home dir/path for dash.
-        # All init and segment files will write under this dir.
-        dash_path           ./objs/nginx/html;
-        # The DASH MPD file path.
-        # We supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        # Default: [app]/[stream].mpd
-        dash_mpd_file       [app]/[stream].mpd;
-    }
-}
-
-# the vhost with hls specified.
-vhost hls.srs.com {
-    hls {
-        # whether the hls is enabled.
-        # if off, do not write hls(ts and m3u8) when publish.
-        # default: off
-        enabled         on;
-        # the hls fragment in seconds, the duration of a piece of ts.
-        # default: 10
-        hls_fragment    10;
-        # the hls m3u8 target duration ratio,
-        #   EXT-X-TARGETDURATION = hls_td_ratio * hls_fragment // init
-        #   EXT-X-TARGETDURATION = max(ts_duration, EXT-X-TARGETDURATION) // for each ts
-        # @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081
-        # default: 1.5
-        hls_td_ratio    1.5;
-        # the audio overflow ratio.
-        # for pure audio, the duration to reap the segment.
-        # for example, the hls_fragment is 10s, hls_aof_ratio is 2.0,
-        # the segment will reap to 20s for pure audio.
-        # default: 2.0
-        hls_aof_ratio   2.0;
-        # the hls window in seconds, the number of ts in m3u8.
-        # default: 60
-        hls_window      60;
-        # the error strategy. can be:
-        #       ignore, disable the hls.
-        #       disconnect, require encoder republish.
-        #       continue, ignore failed try to continue output hls.
-        # @see https://github.com/ossrs/srs/issues/264
-        # default: continue
-        hls_on_error    continue;
-        # the hls output path.
-        # the m3u8 file is configured by hls_path/hls_m3u8_file, the default is:
-        #       ./objs/nginx/html/[app]/[stream].m3u8
-        # the ts file is configured by hls_path/hls_ts_file, the default is:
-        #       ./objs/nginx/html/[app]/[stream]-[seq].ts
-        # @remark the hls_path is compatible with srs v1 config.
-        # default: ./objs/nginx/html
-        hls_path        ./objs/nginx/html;
-        # the hls m3u8 file name.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        # default: [app]/[stream].m3u8
-        hls_m3u8_file   [app]/[stream].m3u8;
-        # the hls ts file name.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        #       [2006], replace this const to current year.
-        #       [01], replace this const to current month.
-        #       [02], replace this const to current date.
-        #       [15], replace this const to current hour.
-        #       [04], replace this const to current minute.
-        #       [05], replace this const to current second.
-        #       [999], replace this const to current millisecond.
-        #       [timestamp],replace this const to current UNIX timestamp in ms.
-        #       [seq], the sequence number of ts.
-        #       [duration], replace this const to current ts duration.
-        # @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
-        # @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
-        # default: [app]/[stream]-[seq].ts
-        hls_ts_file     [app]/[stream]-[seq].ts;
-        # whether use floor for the hls_ts_file path generation.
-        # if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
-        #       and use enhanced algorithm to calc deviation for segment.
-        # @remark when floor on, recommend the hls_segment>=2*gop.
-        # default: off
-        hls_ts_floor    off;
-        # the hls entry prefix, which is base url of ts url.
-        # for example, the prefix is:
-        #         http://your-server/
-        # then, the ts path in m3u8 will be like:
-        #         http://your-server/live/livestream-0.ts
-        #         http://your-server/live/livestream-1.ts
-        #         ...
-        # optional, default to empty string.
-        hls_entry_prefix http://your-server;
-        # the default audio codec of hls.
-        # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
-        # so user can set the default codec for mp3.
-        # the available audio codec:
-        #       aac, mp3, an
-        # default: aac
-        hls_acodec      aac;
-        # the default video codec of hls.
-        # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
-        # so user can set the default codec for pure audio(without video) to vn.
-        # the available video codec:
-        #       h264, vn
-        # default: h264
-        hls_vcodec      h264;
-        # whether cleanup the old expired ts files.
-        # default: on
-        hls_cleanup     on;
-        # the timeout in seconds to dispose the hls,
-        # dispose is to remove all hls files, m3u8 and ts files.
-        # when publisher timeout dispose hls.
-        # @remark 0 to disable dispose for publisher.
-        # @remark apply for publisher timeout only, while "etc/init.d/srs stop" always dispose hls.
-        # default: 0
-        hls_dispose     0;
-        # the max size to notify hls,
-        # to read max bytes from ts of specified cdn network,
-        # @remark only used when on_hls_notify is config.
-        # default: 64
-        hls_nb_notify   64;
-        # whether wait keyframe to reap segment,
-        # if off, reap segment when duration exceed the fragment,
-        # if on, reap segment when duration exceed and got keyframe.
-        # default: on
-        hls_wait_keyframe       on;
-
-        # whether using AES encryption.
-        # default: off
-        hls_keys        on; 
-        # the number of clear ts which one key can encrypt.
-        # default: 5
-        hls_fragments_per_key 5;
-        # the hls key file name.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        #       [seq], the sequence number of key corresponding to the ts.
-        hls_key_file     [app]/[stream]-[seq].key;
-        # the key output path.
-        # the key file is configed by hls_path/hls_key_file, the default is:
-        # ./objs/nginx/html/[app]/[stream]-[seq].key
-        hls_key_file_path    ./objs/nginx/html;
-        # the key root URL, use this can support https.
-        # @remark It's optional.
-        hls_key_url       https://localhost:8080;
-
-        # on_hls, never config in here, should config in http_hooks.
-        # for the hls http callback, @see http_hooks.on_hls of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
-
-        # on_hls_notify, never config in here, should config in http_hooks.
-        # we support the variables to generate the notify url:
-        #       [app], replace with the app.
-        #       [stream], replace with the stream.
-        #       [param], replace with the param.
-        #       [ts_url], replace with the ts url.
-        # for the hls http callback, @see http_hooks.on_hls_notify of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#on-hls-notify
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#on-hls-notify
-    }
-}
-# the vhost with hls disabled.
-vhost no-hls.srs.com {
-    hls {
-        # whether the hls is enabled.
-        # if off, do not write hls(ts and m3u8) when publish.
-        # default: off
-        enabled         off;
-    }
-}
-
-# the vhost with adobe hds
-vhost hds.srs.com {
-    hds {
-        # whether hds enabled
-        # default: off
-        enabled         on;
-        # the hds fragment in seconds.
-        # default: 10
-        hds_fragment    10;
-        # the hds window in seconds, erase the segment when exceed the window.
-        # default: 60
-        hds_window      60;
-        # the path to store the hds files.
-        # default: ./objs/nginx/html
-        hds_path        ./objs/nginx/html;
-    }
-}
-
-# vhost for dvr
-vhost dvr.srs.com {
-    # DVR RTMP stream to file,
-    # start to record to file when encoder publish,
-    # reap flv/mp4 according by specified dvr_plan.
-    dvr {
-        # whether enabled dvr features
-        # default: off
-        enabled         on;
-        # the filter for dvr to apply to.
-        #       all, dvr all streams of all apps.
-        #       <app>/<stream>, apply to specified stream of app.
-        # for example, to dvr the following two streams:
-        #       live/stream1 live/stream2
-        # default: all
-        dvr_apply       all;
-        # the dvr plan. canbe:
-        #       session reap flv/mp4 when session end(unpublish).
-        #       segment reap flv/mp4 when flv duration exceed the specified dvr_duration.
-        # @remark The plan append is removed in SRS3+, for it's no use.
-        # default: session
-        dvr_plan        session;
-        # the dvr output path, *.flv or *.mp4.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        #       [2006], replace this const to current year.
-        #       [01], replace this const to current month.
-        #       [02], replace this const to current date.
-        #       [15], replace this const to current hour.
-        #       [04], replace this const to current minute.
-        #       [05], replace this const to current second.
-        #       [999], replace this const to current millisecond.
-        #       [timestamp],replace this const to current UNIX timestamp in ms.
-        # @remark we use golang time format "2006-01-02 15:04:05.999" as "[2006]-[01]-[02]_[15].[04].[05]_[999]"
-        # for example, for url rtmp://ossrs.net/live/livestream and time 2015-01-03 10:57:30.776
-        # 1. No variables, the rule of SRS1.0(auto add [stream].[timestamp].flv as filename):
-        #       dvr_path ./objs/nginx/html;
-        #       =>
-        #       dvr_path ./objs/nginx/html/live/livestream.1420254068776.flv;
-        # 2. Use stream and date as dir name, time as filename:
-        #       dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
-        #       =>
-        #       dvr_path /data/ossrs.net/live/livestream/2015/01/03/10.57.30.776.flv;
-        # 3. Use stream and year/month as dir name, date and time as filename:
-        #       dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]-[15].[04].[05].[999].flv;
-        #       =>
-        #       dvr_path /data/ossrs.net/live/livestream/2015/01/03-10.57.30.776.flv;
-        # 4. Use vhost/app and year/month as dir name, stream/date/time as filename:
-        #       dvr_path /data/[vhost]/[app]/[2006]/[01]/[stream]-[02]-[15].[04].[05].[999].flv;
-        #       =>
-        #       dvr_path /data/ossrs.net/live/2015/01/livestream-03-10.57.30.776.flv;
-        # 5. DVR to mp4:
-        #       dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].mp4;
-        #       =>
-        #       dvr_path ./objs/nginx/html/live/livestream.1420254068776.mp4;
-        # @see https://github.com/ossrs/srs/wiki/v3_CN_DVR#custom-path
-        # @see https://github.com/ossrs/srs/wiki/v3_EN_DVR#custom-path
-        #       segment,session apply it.
-        # default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
-        dvr_path        ./objs/nginx/html/[app]/[stream].[timestamp].flv;
-        # the duration for dvr file, reap if exceed, in seconds.
-        #       segment apply it.
-        #       session,append ignore.
-        # default: 30
-        dvr_duration    30;
-        # whether wait keyframe to reap segment,
-        # if off, reap segment when duration exceed the dvr_duration,
-        # if on, reap segment when duration exceed and got keyframe.
-        #       segment apply it.
-        #       session,append ignore.
-        # default: on
-        dvr_wait_keyframe       on;
-        # about the stream monotonically increasing:
-        #   1. video timestamp is monotonically increasing,
-        #   2. audio timestamp is monotonically increasing,
-        #   3. video and audio timestamp is interleaved monotonically increasing.
-        # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
-        # however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
-        # the time jitter algorithm:
-        #   1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
-        #   2. zero, only ensure stream start at zero, ignore timestamp jitter.
-        #   3. off, disable the time jitter algorithm, like atc.
-        # apply for all dvr plan.
-        # default: full
-        time_jitter             full;
-
-        # on_dvr, never config in here, should config in http_hooks.
-        # for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DVR#http-callback
-    }
-}
-
-# vhost for ingest
-vhost ingest.srs.com {
-    # ingest file/stream/device then push to SRS over RTMP.
-    # the name/id used to identify the ingest, must be unique in global.
-    # ingest id is used in reload or http api management.
-    # @remark vhost can contains multiple ingest
-    ingest livestream {
-        # whether enabled ingest features
-        # default: off
-        enabled      on;
-        # input file/stream/device
-        # @remark only support one input.
-        input {
-            # the type of input.
-            # can be file/stream/device, that is,
-            #   file: ingest file specified by url.
-            #   stream: ingest stream specified by url.
-            #   device: not support yet.
-            # default: file
-            type    file;
-            # the url of file/stream.
-            url     ./doc/source.200kbps.768x320.flv;
-        }
-        # the ffmpeg
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        # the transcode engine, @see all.transcode.srs.com
-        # @remark, the output is specified following.
-        engine {
-            # @see enabled of transcode engine.
-            # if disabled or vcodec/acodec not specified, use copy.
-            # default: off.
-            enabled          off;
-            # output stream. variables:
-            #       [vhost] current vhost which start the ingest.
-            #       [port] system RTMP stream port.
-            output          rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
-        }
-    }
-}
-
-# the vhost for ingest with transcode engine.
-vhost transcode.ingest.srs.com {
-    ingest livestream {
-        enabled      on;
-        input {
-            type    file;
-            url     ./doc/source.200kbps.768x320.flv;
-        }
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine {
-            enabled         off;
-            perfile {
-                re;
-                rtsp_transport tcp;
-            }
-            iformat         flv;
-            vfilter {
-                i               ./doc/ffmpeg-logo.png;
-                filter_complex  'overlay=10:10';
-            }
-            vcodec          libx264;
-            vbitrate        1500;
-            vfps            25;
-            vwidth          768;
-            vheight         320;
-            vthreads        12;
-            vprofile        main;
-            vpreset         medium;
-            vparams {
-                t               100;
-                coder           1;
-                b_strategy      2;
-                bf              3;
-                refs            10;
-            }
-            acodec          libfdk_aac;
-            abitrate        70;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-                profile:a   aac_low;
-            }
-            oformat         flv;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream];
-        }
-    }
-}
-
-# the main comments for transcode
-vhost example.transcode.srs.com {
-    # the streaming transcode configs.
-    # @remark vhost can contains multiple transcode
-    transcode {
-        # whether the transcode enabled.
-        # if off, donot transcode.
-        # default: off.
-        enabled     on;
-        # the ffmpeg
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        # the transcode engine for matched stream.
-        # all matched stream will transcoded to the following stream.
-        # the transcode set name(ie. hd) is optional and not used.
-        # we will build the parameters to fork ffmpeg:
-        #       ffmpeg <perfile>
-        #               -i <iformat> 
-        #               <vfilter> 
-        #               -vcodec <vcodec> -b:v <vbitrate> -r <vfps> -s <vwidth>x<vheight> -profile:v <vprofile> -preset <vpreset>
-        #               <vparams>
-        #               -acodec <acodec> -b:a <abitrate> -ar <asample_rate> -ac <achannels>
-        #               <aparams>
-        #               -f <oformat>
-        #               -y <output>
-        engine example {
-            # whether the engine is enabled
-            # default: off.
-            enabled         on;
-            # pre-file options, before "-i"
-            perfile {
-                re;
-                rtsp_transport tcp;
-            }
-            # input format "-i", can be:
-            #       off, do not specifies the format, ffmpeg will guess it.
-            #       flv, for flv or RTMP stream.
-            #       other format, for example, mp4/aac whatever.
-            # default: flv
-            iformat         flv;
-            # ffmpeg filters, between "-i" and "-vcodec"
-            # follows the main input.
-            vfilter {
-                # the logo input file.
-                i               ./doc/ffmpeg-logo.png;
-                # the ffmpeg complex filter.
-                # for filters, @see: http://ffmpeg.org/ffmpeg-filters.html
-                filter_complex  'overlay=10:10';
-            }
-            # video encoder name, "ffmpeg -vcodec"
-            # can be:
-            #       libx264: use h.264(libx264) video encoder.
-            #       png: use png to snapshot thumbnail.
-            #       copy: donot encoder the video stream, copy it.
-            #       vn: disable video output.
-            vcodec          libx264;
-            # video bitrate, in kbps, "ffmepg -b:v"
-            # @remark 0 to use source video bitrate.
-            # default: 0
-            vbitrate        1500;
-            # video framerate, "ffmepg -r"
-            # @remark 0 to use source video fps.
-            # default: 0
-            vfps            25;
-            # video width, must be even numbers, "ffmepg -s"
-            # @remark 0 to use source video width.
-            # default: 0
-            vwidth          768;
-            # video height, must be even numbers, "ffmepg -s"
-            # @remark 0 to use source video height.
-            # default: 0
-            vheight         320;
-            # the max threads for ffmpeg to used, "ffmepg -thread"
-            # default: 1
-            vthreads        12;
-            # x264 profile, "ffmepg -profile:v"
-            # @see x264 -help, can be:
-            # high,main,baseline
-            vprofile        main;
-            # x264 preset, "ffmpeg -preset"
-            # @see x264 -help, can be:
-            #       ultrafast,superfast,veryfast,faster,fast
-            #       medium,slow,slower,veryslow,placebo
-            vpreset         medium;
-            # other x264 or ffmpeg video params, between "-preset" and "-acodec"
-            vparams {
-                # ffmpeg options, @see: http://ffmpeg.org/ffmpeg.html
-                t               100;
-                # 264 params, @see: http://ffmpeg.org/ffmpeg-codecs.html#libx264
-                coder           1;
-                b_strategy      2;
-                bf              3;
-                refs            10;
-            }
-            # audio encoder name, "ffmpeg -acodec"
-            # can be:
-            #       libfdk_aac: use aac(libfdk_aac) audio encoder.
-            #       copy: donot encoder the audio stream, copy it.
-            #       an: disable audio output.
-            acodec          libfdk_aac;
-            # audio bitrate, in kbps, "ffmpeg -b:a"
-            # [16, 72] for libfdk_aac.
-            # @remark 0 to use source audio bitrate.
-            # default: 0
-            abitrate        70;
-            # audio sample rate, "ffmpeg -ar"
-            # for flv/rtmp, it must be:
-            #       44100,22050,11025,5512
-            # @remark 0 to use source audio sample rate.
-            # default: 0
-            asample_rate    44100;
-            # audio channel, "ffmpeg -ac"
-            # 1 for mono, 2 for stereo.
-            # @remark 0 to use source audio channels.
-            # default: 0
-            achannels       2;
-            # other ffmpeg audio params, between "-ac" and "-f"/"-y"
-            aparams {
-                # audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
-                # @remark SRS supported aac profile for HLS is: aac_low, aac_he, aac_he_v2
-                profile:a   aac_low;
-                bsf:a       aac_adtstoasc;
-            }
-            # output format, "ffmpeg -f" can be:
-            #       off, do not specifies the format, ffmpeg will guess it.
-            #       flv, for flv or RTMP stream.
-            #       image2, for vcodec png to snapshot thumbnail.
-            #       other format, for example, mp4/aac whatever.
-            # default: flv
-            oformat         flv;
-            # output stream, "ffmpeg -y", variables:
-            #       [vhost] the input stream vhost.
-            #       [port] the input stream port.
-            #       [app] the input stream app.
-            #       [stream] the input stream name.
-            #       [engine] the transcode engine name.
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
-vhost mirror.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine mirror {
-            enabled         on;
-            vfilter {
-                vf                  'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
-            }
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# the drawtext filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
-# remark: we remove the libfreetype which always cause build failed, you must add it manual if needed.
-#######################################################################################################
-# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
-vhost crop.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine crop {
-            enabled         on;
-            vfilter {
-                vf                  'crop=in_w-20:in_h-160:10:80';
-            }
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# the logo filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#overlay
-vhost logo.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine logo {
-            enabled         on;
-            vfilter {
-                i               ./doc/ffmpeg-logo.png;
-                filter_complex      'overlay=10:10';
-            }
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# audio transcode only.
-# for example, FMLE publish audio codec in mp3, and do not support HLS output,
-# we can transcode the audio to aac and copy video to the new stream with HLS.
-vhost audio.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine acodec {
-            enabled         on;
-            vcodec          copy;
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# disable video, transcode/copy audio.
-# for example, publish pure audio stream.
-vhost vn.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine vn {
-            enabled         on;
-            vcodec          vn;
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# ffmpeg-copy(forward implements by ffmpeg).
-# copy the video and audio to a new stream.
-vhost copy.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine copy {
-            enabled         on;
-            vcodec          copy;
-            acodec          copy;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# transcode all app and stream of vhost
-# the comments, read example.transcode.srs.com
-vhost all.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine ffsuper {
-            enabled         on;
-            iformat         flv;
-            vfilter {
-                i               ./doc/ffmpeg-logo.png;
-                filter_complex  'overlay=10:10';
-            }
-            vcodec          libx264;
-            vbitrate        1500;
-            vfps            25;
-            vwidth          768;
-            vheight         320;
-            vthreads        12;
-            vprofile        main;
-            vpreset         medium;
-            vparams {
-                t               100;
-                coder           1;
-                b_strategy      2;
-                bf              3;
-                refs            10;
-            }
-            acodec          libfdk_aac;
-            abitrate        70;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-                profile:a   aac_low;
-            }
-            oformat         flv;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine ffhd {
-            enabled         on;
-            vcodec          libx264;
-            vbitrate        1200;
-            vfps            25;
-            vwidth          1382;
-            vheight         576;
-            vthreads        6;
-            vprofile        main;
-            vpreset         medium;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        70;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine ffsd {
-            enabled         on;
-            vcodec          libx264;
-            vbitrate        800;
-            vfps            25;
-            vwidth          1152;
-            vheight         480;
-            vthreads        4;
-            vprofile        main;
-            vpreset         fast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        60;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine fffast {
-            enabled     on;
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine vcopy {
-            enabled         on;
-            vcodec          copy;
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine acopy {
-            enabled     on;
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          copy;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine copy {
-            enabled         on;
-            vcodec          copy;
-            acodec          copy;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# transcode all stream using the empty ffmpeg demo, do nothing.
-vhost ffempty.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg ./objs/research/ffempty;
-        engine empty {
-            enabled         on;
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# transcode all app and stream of app
-vhost app.transcode.srs.com {
-    # the streaming transcode configs.
-    # if app specified, transcode all streams of app.
-    transcode live {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine {
-            enabled     off;
-        }
-    }
-}
-# transcode specified stream.
-vhost stream.transcode.srs.com {
-    # the streaming transcode configs.
-    # if stream specified, transcode the matched stream.
-    transcode live/livestream {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine {
-            enabled     off;
-        }
-    }
-}
-
-#############################################################################################
-# The origin cluster section
-#############################################################################################
-http_api {
-    enabled         on;
-    listen          9090;
-}
-vhost a.origin.cluster.srs.com {
-    cluster {
-        mode                local;
-        origin_cluster      on;
-        coworkers           127.0.0.1:9091;
-    }
-}
-
-http_api {
-    enabled         on;
-    listen          9091;
-}
-vhost b.origin.cluster.srs.com {
-    cluster {
-        mode                local;
-        origin_cluster      on;
-        coworkers           127.0.0.1:9090;
-    }
-}
diff --git a/publish/src/linux-x64/publish/apps/srs/conf/full.conf b/publish/src/linux-x64/publish/apps/srs/conf/full.conf
deleted file mode 100644
index cf3e0217..00000000
--- a/publish/src/linux-x64/publish/apps/srs/conf/full.conf
+++ /dev/null
@@ -1,1848 +0,0 @@
-# all config for srs
-
-#############################################################################################
-# RTMP sections
-#############################################################################################
-# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
-# for example, 192.168.1.100:1935 10.10.10.100:1935
-# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
-listen              1935;
-# the pid file
-# to ensure only one process can use a pid file
-# and provides the current running process id, for script,
-# for example, init.d script to manage the server.
-# default: ./objs/srs.pid
-pid                 ./objs/srs.pid;
-# the default chunk size is 128, max is 65536,
-# some client does not support chunk size change,
-# however, most clients support it and it can improve
-# performance about 10%.
-# default: 60000
-chunk_size          60000;
-# the logs dir.
-# if enabled ffmpeg, each transcoding stream will create a log file.
-# /dev/null to disable the log.
-# default: ./objs
-ff_log_dir          ./objs;
-# the log tank, console or file.
-# if console, print log to console.
-# if file, write log to file. requires srs_log_file if log to file.
-# default: file.
-srs_log_tank        file;
-# the log level, for all log tanks.
-# can be: verbose, info, trace, warn, error
-# default: trace
-srs_log_level       trace;
-# when srs_log_tank is file, specifies the log file.
-# default: ./objs/srs.log
-srs_log_file        ./objs/srs.log;
-# the max connections.
-# if exceed the max connections, server will drop the new connection.
-# default: 1000
-max_connections     1000;
-# whether start as daemon
-# @remark: do not support reload.
-# default: on
-daemon              on;
-# whether use utc_time to generate the time struct,
-# if off, use localtime() to generate it,
-# if on, use gmtime() instead, which use UTC time.
-# default: off
-utc_time            off;
-# config for the pithy print in ms,
-# which always print constant message specified by interval,
-# whatever the clients in concurrency.
-# default: 10000
-pithy_print_ms      10000;
-
-# the work dir for server, to chdir(work_dir) when not empty or "./"
-# user can config this directory to change the dir.
-# @reamrk do not support reload.
-# default: ./
-work_dir ./;
-# whether quit when parent process changed,
-# used for supervisor mode(not daemon), srs should always quit when 
-# supervisor process exited.
-# @remark conflict with daemon, error when both daemon and asprocess are on.
-# @reamrk do not support reload.
-# default: off
-asprocess off;
-
-#############################################################################################
-# heartbeat/stats sections
-#############################################################################################
-# heartbeat to api server
-# @remark, the ip report to server, is retrieve from system stat,
-#       which need the config item stats.network.
-heartbeat {
-    # whether heartbeat is enabled.
-    # default: off
-    enabled         off;
-    # the interval seconds for heartbeat,
-    # recommend 0.3,0.6,0.9,1.2,1.5,1.8,2.1,2.4,2.7,3,...,6,9,12,....
-    # default: 9.9
-    interval        9.3;
-    # when startup, srs will heartbeat to this api.
-    # @remark: must be a restful http api url, where SRS will POST with following data:
-    #   {
-    #       "device_id": "my-srs-device",
-    #       "ip": "192.168.1.100"
-    #   }
-    # default: http://127.0.0.1:8085/api/v1/servers
-    url             http://127.0.0.1:8085/api/v1/servers;
-    # the id of device.
-    device_id       "my-srs-device";
-    # whether report with summaries
-    # if on, put /api/v1/summaries to the request data:
-    #   {
-    #       "summaries": summaries object.
-    #   }
-    # @remark: optional config.
-    # default: off
-    summaries       off;
-}
-
-# system statistics section.
-# the main cycle will retrieve the system stat,
-# for example, the cpu/mem/network/disk-io data,
-# the http api, for instance, /api/v1/summaries will show these data.
-# @remark the heartbeat depends on the network,
-#       for example, the eth0 maybe the device which index is 0.
-stats {
-    # the index of device ip.
-    # we may retrieve more than one network device.
-    # default: 0
-    network         0;
-    # the device name to stat the disk iops.
-    # ignore the device of /proc/diskstats if not configured.
-    disk            sda sdb xvda xvdb;
-}
-
-#############################################################################################
-# HTTP sections
-#############################################################################################
-# api of srs.
-# the http api config, export for external program to manage srs.
-# user can access http api of srs in browser directly, for instance, to access by:
-#       curl http://192.168.1.170:1985/api/v1/reload
-# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
-# where the cli can only be used in shell/terminate.
-http_api {
-    # whether http api is enabled.
-    # default: off
-    enabled         on;
-    # the http api listen entry is <[ip:]port>
-    # for example, 192.168.1.100:1985
-    # where the ip is optional, default to 0.0.0.0, that is 1985 equals to 0.0.0.0:1985
-    # default: 1985
-    listen          1985;
-    # whether enable crossdomain request.
-    # default: on
-    crossdomain     on;
-    # the HTTP RAW API is more powerful api to change srs state and reload.
-    raw_api {
-        # whether enable the HTTP RAW API.
-        # default: off
-        enabled             off;
-        # whether enable rpc reload.
-        # default: off
-        allow_reload        off;
-        # whether enable rpc query.
-        # default: off
-        allow_query         off;
-        # whether enable rpc update.
-        # default: off
-        allow_update        off;
-    }
-}
-# embedded http server in srs.
-# the http streaming config, for HLS/HDS/DASH/HTTPProgressive
-# global config for http streaming, user must config the http section for each vhost.
-# the embed http server used to substitute nginx in ./objs/nginx,
-# for example, srs running in arm, can provides RTMP and HTTP service, only with srs installed.
-# user can access the http server pages, generally:
-#       curl http://192.168.1.170:80/srs.html
-# which will show srs version and welcome to srs.
-# @remark, the http embedded stream need to config the vhost, for instance, the __defaultVhost__
-# need to open the feature http of vhost.
-http_server {
-    # whether http streaming service is enabled.
-    # default: off
-    enabled         on;
-    # the http streaming listen entry is <[ip:]port>
-    # for example, 192.168.1.100:8080
-    # where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
-    # @remark, if use lower port, for instance 80, user must start srs by root.
-    # default: 8080
-    listen          8080;
-    # the default dir for http root.
-    # default: ./objs/nginx/html
-    dir             ./objs/nginx/html;
-    # whether enable crossdomain request.
-    # for both http static and stream server and apply on all vhosts.
-    # default: on
-    crossdomain     on;
-}
-
-#############################################################################################
-# Streamer sections
-#############################################################################################
-# the streamer cast stream from other protocol to SRS over RTMP.
-# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
-stream_caster {
-    # whether stream caster is enabled.
-    # default: off
-    enabled         off;
-    # the caster type of stream, the casters:
-    #       mpegts_over_udp, MPEG-TS over UDP caster.
-    #       rtsp, Real Time Streaming Protocol (RTSP).
-    #       flv, FLV over HTTP POST.
-    caster          mpegts_over_udp;
-    # the output rtmp url.
-    # for mpegts_over_udp caster, the typically output url:
-    #           rtmp://127.0.0.1/live/livestream
-    # for rtsp caster, the typically output url:
-    #           rtmp://127.0.0.1/[app]/[stream]
-    #       for example, the rtsp url:
-    #           rtsp://192.168.1.173:8544/live/livestream.sdp
-    #       where the [app] is "live" and [stream] is "livestream", output is:
-    #           rtmp://127.0.0.1/live/livestream
-    # for flv caster, the typically output url:
-    #           rtmp://127.0.0.1/[app]/[stream]
-    #       for example, POST to url:
-    #           http://127.0.0.1:8936/live/livestream.flv
-    #       where the [app] is "live" and [stream] is "livestream", output is:
-    #           rtmp://127.0.0.1/live/livestream
-    output          rtmp://127.0.0.1/live/livestream;
-    # the listen port for stream caster.
-    #       for mpegts_over_udp caster, listen at udp port. for example, 8935.
-    #       for rtsp caster, listen at tcp port. for example, 554.
-    #       for flv caster, listen at tcp port. for example, 8936.
-    # TODO: support listen at <[ip:]port>
-    listen          8935;
-    # for the rtsp caster, the rtp server local port over udp,
-    # which reply the rtsp setup request message, the port will be used:
-    #       [rtp_port_min, rtp_port_max)
-    rtp_port_min    57200;
-    rtp_port_max    57300;
-}
-stream_caster {
-    enabled         off;
-    caster          mpegts_over_udp;
-    output          rtmp://127.0.0.1/live/livestream;
-    listen          8935;
-}
-stream_caster {
-    enabled         off;
-    caster          rtsp;
-    output          rtmp://127.0.0.1/[app]/[stream];
-    listen          554;
-    rtp_port_min    57200;
-    rtp_port_max    57300;
-}
-stream_caster {
-    enabled         off;
-    caster          flv;
-    output          rtmp://127.0.0.1/[app]/[stream];
-    listen          8936;
-}
-
-#############################################################################################
-# Kafka sections
-#############################################################################################
-# Apache Kafka is a high-throughput distributed messaging system.
-# SRS is a Kafka producer to send message to kafka.
-# @see https://kafka.apache.org/documentation.html#introduction
-kafka {
-    # whether enabled kafka.
-    # default: off
-    enabled         off;
-    # the broker list, broker is <ip:port>
-    # and use space to specify multple brokers.
-    # for example, 127.0.0.1:9092 127.0.0.1:9093
-    brokers         127.0.0.1:9092;
-    # the kafka topic to use.
-    # default: srs
-    topic           srs;
-}
-
-#############################################################################################
-# RTMP/HTTP VHOST sections
-#############################################################################################
-# vhost list, the __defaultVhost__ is the default vhost
-# for example, user use ip to access the stream: rtmp://192.168.1.2/live/livestream.
-# for which cannot identify the required vhost.
-vhost __defaultVhost__ {
-}
-
-# the vhost scope configs.
-vhost scope.vhost.srs.com {
-    # whether the vhost is enabled.
-    # if off, all request access denied.
-    # default: on
-    enabled         off;
-
-    # whether enable min delay mode for vhost.
-    # for min latency mode:
-    # 1. disable the publish.mr for vhost.
-    # 2. use timeout for cond wait for consumer queue.
-    # @see https://github.com/ossrs/srs/issues/257
-    # default: off
-    min_latency     off;
-
-    # whether enable the TCP_NODELAY
-    # if on, set the nodelay of fd by setsockopt
-    # default: off
-    tcp_nodelay     off;
-
-    # the default chunk size is 128, max is 65536,
-    # some client does not support chunk size change,
-    # vhost chunk size will override the global value.
-    # default: global chunk size.
-    chunk_size      128;
-    
-    # The input ack size, 0 to not set.
-    # Generally, it's set by the message from peer,
-    # but for some peer(encoder), it never send message but use a different ack size.
-    # We can chnage the default ack size in server-side, to send acknowledge message,
-    # or the encoder maybe blocked after publishing for some time.
-    # Default: 0
-    in_ack_size     0;
-    
-    # The output ack size, 0 to not set.
-    # This is used to notify the peer(player) to send acknowledge to server.
-    # Default: 2500000
-    out_ack_size    2500000;
-}
-
-# set the chunk size of vhost.
-vhost chunksize.srs.com {
-    # @see scope.vhost.srs.com
-    chunk_size      128;
-}
-
-# the vhost disabled.
-vhost removed.srs.com {
-    # @see scope.vhost.srs.com
-    enabled         off;
-}
-
-# vhost for stream cluster for RTMP/FLV
-vhost cluster.srs.com {
-    # The config for cluster.
-    cluster {
-        # The cluster mode, local or remote.
-        #       local: It's an origin server, serve streams itself.
-        #       remote: It's an edge server, fetch or push stream to origin server.
-        # default: local
-        mode            remote;
-
-        # For edge(mode remote), user must specifies the origin server
-        # format as: <server_name|ip>[:port]
-        # @remark user can specifies multiple origin for error backup, by space,
-        # for example, 192.168.1.100:1935 192.168.1.101:1935 192.168.1.102:1935
-        origin          127.0.0.1:1935 localhost:1935;
-
-        # For edge(mode remote), whether open the token traverse mode,
-        # if token traverse on, all connections of edge will forward to origin to check(auth),
-        # it's very important for the edge to do the token auth.
-        # the better way is use http callback to do the token auth by the edge,
-        # but if user prefer origin check(auth), the token_traverse if better solution.
-        # default: off
-        token_traverse  off;
-
-        # For edge(mode remote), the vhost to transform for edge,
-        # to fetch from the specified vhost at origin,
-        # if not specified, use the current vhost of edge in origin, the variable [vhost].
-        # default: [vhost]
-        vhost           same.edge.srs.com;
-
-        # For edge(mode remote), when upnode(forward to, edge push to, edge pull from) is srs,
-        # it's strongly recommend to open the debug_srs_upnode,
-        # when connect to upnode, it will take the debug info,
-        # for example, the id, source id, pid.
-        # please see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLog
-        # default: on
-        debug_srs_upnode    on;
-
-        # For origin(mode local) cluster, turn on the cluster.
-        # @remark Origin cluster only supports RTMP, use Edge to transmux RTMP to FLV.
-        # default: off
-        # TODO: FIXME: Support reload.
-        origin_cluster      off;
-
-        # For origin (mode local) cluster, the co-worker's HTTP APIs.
-        # This origin will connect to co-workers and communicate with them.
-        # please read: https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
-        # TODO: FIXME: Support reload.
-        coworkers           127.0.0.1:9091 127.0.0.1:9092;
-    }
-}
-
-# vhost for edge, edge and origin is the same vhost
-vhost same.edge.srs.com {
-    # @see cluster.srs.com
-    cluster {
-        mode            remote;
-        origin          127.0.0.1:1935 localhost:1935;
-        token_traverse  off;
-    }
-}
-
-# vhost for edge, edge transform vhost to fetch from another vhost.
-vhost transform.edge.srs.com {
-    # @see cluster.srs.com
-    cluster {
-        mode            remote;
-        origin          127.0.0.1:1935;
-        vhost           same.edge.srs.com;
-    }
-}
-
-# the vhost for srs debug info, whether send args in connect(tcUrl).
-vhost debug.srs.com {
-    # @see cluster.srs.com
-    cluster {
-        debug_srs_upnode    on;
-    }
-}
-
-# the vhost which forward publish streams.
-vhost same.vhost.forward.srs.com {
-    # forward stream to other servers.
-    forward {
-        # whether enable the forward.
-        # default: off
-        enabled on;
-        # forward all publish stream to the specified server.
-        # this used to split/forward the current stream for cluster active-standby,
-        # active-active for cdn to build high available fault tolerance system.
-        # format: {ip}:{port} {ip_N}:{port_N}
-        destination 127.0.0.1:1936 127.0.0.1:1937;
-    }
-}
-
-# the play specified configs
-vhost play.srs.com {
-    # for play client, both RTMP and other stream clients,
-    # for instance, the HTTP FLV stream clients.
-    play {
-        # whether cache the last gop.
-        # if on, cache the last gop and dispatch to client,
-        #   to enabled fast startup for client, client play immediately.
-        # if off, send the latest media data to client,
-        #   client need to wait for the next Iframe to decode and show the video.
-        # set to off if requires min delay;
-        # set to on if requires client fast startup.
-        # default: on
-        gop_cache       off;
-        # the max live queue length in seconds.
-        # if the messages in the queue exceed the max length,
-        # drop the old whole gop.
-        # default: 30
-        queue_length    10;
-
-        # about the stream monotonically increasing:
-        #   1. video timestamp is monotonically increasing,
-        #   2. audio timestamp is monotonically increasing,
-        #   3. video and audio timestamp is interleaved/mixed monotonically increasing.
-        # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
-        # however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
-        # the time jitter algorithm:
-        #   1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
-        #   2. zero, only ensure stream start at zero, ignore timestamp jitter.
-        #   3. off, disable the time jitter algorithm, like atc.
-        # @remark for full, correct timestamp only when |delta| > 250ms.
-        # @remark disabled when atc is on.
-        # default: full
-        time_jitter             full;
-        # vhost for atc for hls/hds/rtmp backup.
-        # generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0.
-        # when atc is on, server delivery rtmp stream by absolute time.
-        # atc is used, for instance, encoder will copy stream to master and slave server,
-        # server use atc to delivery stream to edge/client, where stream time from master/slave server
-        # is always the same, client/tools can slice RTMP stream to HLS according to the same time,
-        # if the time not the same, the HLS stream cannot slice to support system backup.
-        #
-        # @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html
-        # @see http://www.baidu.com/#wd=hds%20hls%20atc
-        #
-        # @remark when atc is on, auto off the time_jitter
-        # default: off
-        atc             off;
-        # whether use the interleaved/mixed algorithm to correct the timestamp.
-        # if on, always ensure the timestamp of audio+video is interleaved/mixed monotonically increase.
-        # if off, use time_jitter to correct the timestamp if required.
-        # @remark to use mix_correct, atc should on(or time_jitter should off).
-        # default: off
-        mix_correct             off;
-
-        # whether enable the auto atc,
-        # if enabled, detect the bravo_atc="true" in onMetaData packet,
-        # set atc to on if matched.
-        # always ignore the onMetaData if atc_auto is off.
-        # default: off
-        atc_auto        off;
-
-        # set the MW(merged-write) latency in ms.
-        # SRS always set mw on, so we just set the latency value.
-        # the latency of stream >= mw_latency + mr_latency
-        # the value recomment is [300, 1800]
-        # default: 350
-        mw_latency      350;
-
-        # the minimal packets send interval in ms,
-        # used to control the ndiff of stream by srs_rtmp_dump,
-        # for example, some device can only accept some stream which
-        # delivery packets in constant interval(not cbr).
-        # @remark 0 to disable the minimal interval.
-        # @remark >0 to make the srs to send message one by one.
-        # @remark user can get the right packets interval in ms by srs_rtmp_dump.
-        # default: 0
-        send_min_interval       10.0;
-        # whether reduce the sequence header,
-        # for some client which cannot got duplicated sequence header,
-        # while the sequence header is not changed yet.
-        # default: off
-        reduce_sequence_header  on;
-    }
-}
-
-# vhost for time jitter
-vhost jitter.srs.com {
-    # @see play.srs.com
-    # to use time_jitter full, the default config.
-    play {
-    }
-    # to use mix_correct.
-    play {
-        time_jitter             off;
-        mix_correct             on;
-    }
-    play {
-        atc                     on;
-        mix_correct             on;
-    }
-    # to use atc
-    play {
-        atc                     on;
-    }
-}
-
-# vhost for atc.
-vhost atc.srs.com {
-    # @see play.srs.com
-    play {
-        atc             on;
-        atc_auto        on;
-    }
-}
-
-# the MR(merged-read) setting for publisher.
-# the MW(merged-write) settings for player.
-vhost mrw.srs.com {
-    # @see scope.vhost.srs.com
-    min_latency     off;
-
-    # @see play.srs.com
-    play {
-        mw_latency      350;
-    }
-
-    # @see publish.srs.com
-    publish {
-        mr          on;
-        mr_latency  350;
-    }
-}
-
-# the vhost for min delay, do not cache any stream.
-vhost min.delay.com {
-    # @see scope.vhost.srs.com
-    min_latency     on;
-    # @see scope.vhost.srs.com
-    tcp_nodelay     on;
-
-    # @see play.srs.com
-    play {
-        mw_latency      100;
-        gop_cache       off;
-        queue_length    10;
-    }
-
-    # @see publish.srs.com
-    publish {
-        mr          off;
-    }
-}
-
-# whether disable the sps parse, for the resolution of video.
-vhost no.parse.sps.com {
-    # @see publish.srs.com
-    publish {
-        parse_sps   on;
-    }
-}
-
-# the vhost to control the stream delivery feature
-vhost stream.control.com {
-    # @see scope.vhost.srs.com
-    min_latency     on;
-    # @see scope.vhost.srs.com
-    tcp_nodelay     on;
-
-    # @see play.srs.com
-    play {
-        mw_latency      100;
-        queue_length    10;
-        send_min_interval       10.0;
-        reduce_sequence_header  on;
-    }
-
-    # @see publish.srs.com
-    publish {
-        mr          off;
-        firstpkt_timeout    20000;
-        normal_timeout      7000;
-    }
-}
-
-# the publish specified configs
-vhost publish.srs.com {
-    # the config for FMLE/Flash publisher, which push RTMP to SRS.
-    publish {
-        # about MR, read https://github.com/ossrs/srs/issues/241
-        # when enabled the mr, SRS will read as large as possible.
-        # default: off
-        mr          off;
-        # the latency in ms for MR(merged-read),
-        # the performance+ when latency+, and memory+,
-        #       memory(buffer) = latency * kbps / 8
-        # for example, latency=500ms, kbps=3000kbps, each publish connection will consume
-        #       memory = 500 * 3000 / 8 = 187500B = 183KB
-        # when there are 2500 publisher, the total memory of SRS at least:
-        #       183KB * 2500 = 446MB
-        # the recommended value is [300, 2000]
-        # default: 350
-        mr_latency  350;
-
-        # the 1st packet timeout in ms for encoder.
-        # default: 20000
-        firstpkt_timeout    20000;
-        # the normal packet timeout in ms for encoder.
-        # default: 5000
-        normal_timeout      7000;
-        # whether parse the sps when publish stream.
-        # we can got the resolution of video for stat api.
-        # but we may failed to cause publish failed.
-        # default: on
-        parse_sps   on;
-    }
-}
-
-# the vhost for anti-suck.
-vhost refer.anti_suck.com {
-    # refer hotlink-denial.
-    refer {
-        # whether enable the refer hotlink-denial.
-        # default: off.
-        enabled         on;
-        # the common refer for play and publish.
-        # if the page url of client not in the refer, access denied.
-        # if not specified this field, allow all.
-        # default: not specified.
-        all           github.com github.io;
-        # refer for publish clients specified.
-        # the common refer is not overridden by this.
-        # if not specified this field, allow all.
-        # default: not specified.
-        publish   github.com github.io;
-        # refer for play clients specified.
-        # the common refer is not overridden by this.
-        # if not specified this field, allow all.
-        # default: not specified.
-        play      github.com github.io;
-    }
-}
-
-# vhost for bandwidth check
-# generally, the bandcheck vhost must be: bandcheck.srs.com,
-# or need to modify the vhost of client.
-vhost bandcheck.srs.com {
-    enabled         on;
-    chunk_size      65000;
-    # bandwidth check config.
-    bandcheck {
-        # whether support bandwidth check,
-        # default: off.
-        enabled         on;
-        # the key for server to valid,
-        # if invalid key, server disconnect and abort the bandwidth check.
-        key             "35c9b402c12a7246868752e2878f7e0e";
-        # the interval in seconds for bandwidth check,
-        # server do not allow new test request.
-        # default: 30
-        interval        30;
-        # the max available check bandwidth in kbps.
-        # to avoid attack of bandwidth check.
-        # default: 1000
-        limit_kbps      4000;
-    }
-}
-
-# the security to allow or deny clients.
-vhost security.srs.com {
-    # security for host to allow or deny clients.
-    # @see https://github.com/ossrs/srs/issues/211
-    security {
-        # whether enable the security for vhost.
-        # default: off
-        enabled         on;
-        # the security list, each item format as:
-        #       allow|deny    publish|play    all|<ip>
-        # for example:
-        #       allow           publish     all;
-        #       deny            publish     all;
-        #       allow           publish     127.0.0.1;
-        #       deny            publish     127.0.0.1;
-        #       allow           play        all;
-        #       deny            play        all;
-        #       allow           play        127.0.0.1;
-        #       deny            play        127.0.0.1;
-        # SRS apply the following simple strategies one by one:
-        #       1. allow all if security disabled.
-        #       2. default to deny all when security enabled.
-        #       3. allow if matches allow strategy.
-        #       4. deny if matches deny strategy.
-        allow           play        all;
-        allow           publish     all;
-    }
-}
-
-# vhost for http static and flv vod stream for each vhost.
-vhost http.static.srs.com {
-    # http static vhost specified config
-    http_static {
-        # whether enabled the http static service for vhost.
-        # default: off
-        enabled     on;
-        # the url to mount to,
-        # typical mount to [vhost]/
-        # the variables:
-        #       [vhost] current vhost for http server.
-        # @remark the [vhost] is optional, used to mount at specified vhost.
-        # @remark the http of __defaultVhost__ will override the http_server section.
-        # for example:
-        #       mount to [vhost]/
-        #           access by http://ossrs.net:8080/xxx.html
-        #       mount to [vhost]/hls
-        #           access by http://ossrs.net:8080/hls/xxx.html
-        #       mount to /
-        #           access by http://ossrs.net:8080/xxx.html
-        #           or by http://192.168.1.173:8080/xxx.html
-        #       mount to /hls
-        #           access by http://ossrs.net:8080/hls/xxx.html
-        #           or by http://192.168.1.173:8080/hls/xxx.html
-        # @remark the port of http is specified by http_server section.
-        # default: [vhost]/
-        mount       [vhost]/hls;
-        # main dir of vhost,
-        # to delivery HTTP stream of this vhost.
-        # default: ./objs/nginx/html
-        dir         ./objs/nginx/html/hls;
-    }
-}
-
-# vhost for http flv/aac/mp3 live stream for each vhost.
-vhost http.remux.srs.com {
-    # http flv/mp3/aac/ts stream vhost specified config
-    http_remux {
-        # whether enable the http live streaming service for vhost.
-        # default: off
-        enabled     on;
-        # the fast cache for audio stream(mp3/aac),
-        # to cache more audio and send to client in a time to make android(weixin) happy.
-        # @remark the flv/ts stream ignore it
-        # @remark 0 to disable fast cache for http audio stream.
-        # default: 0
-        fast_cache  30;
-        # the stream mount for rtmp to remux to live streaming.
-        # typical mount to [vhost]/[app]/[stream].flv
-        # the variables:
-        #       [vhost] current vhost for http live stream.
-        #       [app] current app for http live stream.
-        #       [stream] current stream for http live stream.
-        # @remark the [vhost] is optional, used to mount at specified vhost.
-        # the extension:
-        #       .flv mount http live flv stream, use default gop cache.
-        #       .ts mount http live ts stream, use default gop cache.
-        #       .mp3 mount http live mp3 stream, ignore video and audio mp3 codec required.
-        #       .aac mount http live aac stream, ignore video and audio aac codec required.
-        # for example:
-        #       mount to [vhost]/[app]/[stream].flv
-        #           access by http://ossrs.net:8080/live/livestream.flv
-        #       mount to /[app]/[stream].flv
-        #           access by http://ossrs.net:8080/live/livestream.flv
-        #           or by http://192.168.1.173:8080/live/livestream.flv
-        #       mount to [vhost]/[app]/[stream].mp3
-        #           access by http://ossrs.net:8080/live/livestream.mp3
-        #       mount to [vhost]/[app]/[stream].aac
-        #           access by http://ossrs.net:8080/live/livestream.aac
-        #       mount to [vhost]/[app]/[stream].ts
-        #           access by http://ossrs.net:8080/live/livestream.ts
-        # @remark the port of http is specified by http_server section.
-        # default: [vhost]/[app]/[stream].flv
-        mount       [vhost]/[app]/[stream].flv;
-    }
-}
-
-# the http hook callback vhost, srs will invoke the hooks for specified events.
-vhost hooks.callback.srs.com {
-    http_hooks {
-        # whether the http hooks enable.
-        # default off.
-        enabled         on;
-        # when client connect to vhost/app, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_connect",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
-        #           "pageUrl": "http://www.test.com/live.html"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_connect      http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
-        # when client close/disconnect to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_close",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "send_bytes": 10240, "recv_bytes": 10240
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_close        http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
-        # when client(encoder) publish to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_publish",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_publish http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_publish      http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
-        # when client(encoder) stop publish to vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_unpublish",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_unpublish http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_unpublish    http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
-        # when client start to play vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_play",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy",
-        #           "pageUrl": "http://www.test.com/live.html"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_play http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_play         http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
-        # when client stop to play vhost/app/stream, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_stop",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        # support multiple api hooks, format:
-        #       on_stop http://xxx/api0 http://xxx/api1 http://xxx/apiN
-        on_stop         http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
-        # when srs reap a dvr file, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_dvr",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy",
-        #           "cwd": "/usr/local/srs",
-        #           "file": "./objs/nginx/html/live/livestream.1420254068776.flv"
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        on_dvr          http://127.0.0.1:8085/api/v1/dvrs http://localhost:8085/api/v1/dvrs;
-        # when srs reap a ts file of hls, call the hook,
-        # the request in the POST data string is a object encode by json:
-        #       {
-        #           "action": "on_hls",
-        #           "client_id": 1985,
-        #           "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
-        #           "stream": "livestream", "param":"?token=xxx&salt=yyy",
-        #           "duration": 9.36, // in seconds
-        #           "cwd": "/usr/local/srs",
-        #           "file": "./objs/nginx/html/live/livestream/2015-04-23/01/476584165.ts",
-        #           "url": "live/livestream/2015-04-23/01/476584165.ts",
-        #           "m3u8": "./objs/nginx/html/live/livestream/live.m3u8",
-        #           "m3u8_url": "live/livestream/live.m3u8",
-        #           "seq_no": 100
-        #       }
-        # if valid, the hook must return HTTP code 200(Status OK) and response
-        # an int value specifies the error code(0 corresponding to success):
-        #       0
-        on_hls          http://127.0.0.1:8085/api/v1/hls http://localhost:8085/api/v1/hls;
-        # when srs reap a ts file of hls, call this hook,
-        # used to push file to cdn network, by get the ts file from cdn network.
-        # so we use HTTP GET and use the variable following:
-        #       [app], replace with the app.
-        #       [stream], replace with the stream.
-        #       [param], replace with the param.
-        #       [ts_url], replace with the ts url.
-        # ignore any return data of server.
-        # @remark random select a url to report, not report all.
-        on_hls_notify   http://127.0.0.1:8085/api/v1/hls/[app]/[stream]/[ts_url][param];
-    }
-}
-
-# the vhost for exec, fork process when publish stream.
-vhost exec.srs.com {
-    # the exec used to fork process when got some event.
-    exec {
-        # whether enable the exec.
-        # default: off.
-        enabled     off;
-        # when publish stream, exec the process with variables:
-        #       [vhost] the input stream vhost.
-        #       [port] the input stream port.
-        #       [app] the input stream app.
-        #       [stream] the input stream name.
-        #       [engine] the transcode engine name.
-        # other variables for exec only:
-        #       [url] the rtmp url which trigger the publish.
-        #       [tcUrl] the client request tcUrl.
-        #       [swfUrl] the client request swfUrl.
-        #       [pageUrl] the client request pageUrl.
-        # @remark empty to ignore this exec.
-        publish     ./objs/ffmpeg/bin/ffmpeg -f flv -i [url] -c copy -y ./[stream].flv;
-    }
-}
-
-# The vhost for MPEG-DASH.
-vhost dash.srs.com {
-    dash {
-        # Whether DASH is enabled.
-        # Transmux RTMP to DASH if on.
-        # Default: off
-        enabled             on;
-        # The duration of segment in seconds.
-        # Default: 3
-        dash_fragment       3;
-        # The period to update the MPD in seconds.
-        # Default: 30
-        dash_update_period  30;
-        # The depth of timeshift buffer in seconds.
-        # Default: 60
-        dash_timeshift      60;
-        # The base/home dir/path for dash.
-        # All init and segment files will write under this dir.
-        dash_path           ./objs/nginx/html;
-        # The DASH MPD file path.
-        # We supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        # Default: [app]/[stream].mpd
-        dash_mpd_file       [app]/[stream].mpd;
-    }
-}
-
-# the vhost with hls specified.
-vhost hls.srs.com {
-    hls {
-        # whether the hls is enabled.
-        # if off, do not write hls(ts and m3u8) when publish.
-        # default: off
-        enabled         on;
-        # the hls fragment in seconds, the duration of a piece of ts.
-        # default: 10
-        hls_fragment    10;
-        # the hls m3u8 target duration ratio,
-        #   EXT-X-TARGETDURATION = hls_td_ratio * hls_fragment // init
-        #   EXT-X-TARGETDURATION = max(ts_duration, EXT-X-TARGETDURATION) // for each ts
-        # @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081
-        # default: 1.5
-        hls_td_ratio    1.5;
-        # the audio overflow ratio.
-        # for pure audio, the duration to reap the segment.
-        # for example, the hls_fragment is 10s, hls_aof_ratio is 2.0,
-        # the segment will reap to 20s for pure audio.
-        # default: 2.0
-        hls_aof_ratio   2.0;
-        # the hls window in seconds, the number of ts in m3u8.
-        # default: 60
-        hls_window      60;
-        # the error strategy. can be:
-        #       ignore, disable the hls.
-        #       disconnect, require encoder republish.
-        #       continue, ignore failed try to continue output hls.
-        # @see https://github.com/ossrs/srs/issues/264
-        # default: continue
-        hls_on_error    continue;
-        # the hls output path.
-        # the m3u8 file is configured by hls_path/hls_m3u8_file, the default is:
-        #       ./objs/nginx/html/[app]/[stream].m3u8
-        # the ts file is configured by hls_path/hls_ts_file, the default is:
-        #       ./objs/nginx/html/[app]/[stream]-[seq].ts
-        # @remark the hls_path is compatible with srs v1 config.
-        # default: ./objs/nginx/html
-        hls_path        ./objs/nginx/html;
-        # the hls m3u8 file name.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        # default: [app]/[stream].m3u8
-        hls_m3u8_file   [app]/[stream].m3u8;
-        # the hls ts file name.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        #       [2006], replace this const to current year.
-        #       [01], replace this const to current month.
-        #       [02], replace this const to current date.
-        #       [15], replace this const to current hour.
-        #       [04], replace this const to current minute.
-        #       [05], replace this const to current second.
-        #       [999], replace this const to current millisecond.
-        #       [timestamp],replace this const to current UNIX timestamp in ms.
-        #       [seq], the sequence number of ts.
-        #       [duration], replace this const to current ts duration.
-        # @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
-        # @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
-        # default: [app]/[stream]-[seq].ts
-        hls_ts_file     [app]/[stream]-[seq].ts;
-        # whether use floor for the hls_ts_file path generation.
-        # if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
-        #       and use enhanced algorithm to calc deviation for segment.
-        # @remark when floor on, recommend the hls_segment>=2*gop.
-        # default: off
-        hls_ts_floor    off;
-        # the hls entry prefix, which is base url of ts url.
-        # for example, the prefix is:
-        #         http://your-server/
-        # then, the ts path in m3u8 will be like:
-        #         http://your-server/live/livestream-0.ts
-        #         http://your-server/live/livestream-1.ts
-        #         ...
-        # optional, default to empty string.
-        hls_entry_prefix http://your-server;
-        # the default audio codec of hls.
-        # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
-        # so user can set the default codec for mp3.
-        # the available audio codec:
-        #       aac, mp3, an
-        # default: aac
-        hls_acodec      aac;
-        # the default video codec of hls.
-        # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
-        # so user can set the default codec for pure audio(without video) to vn.
-        # the available video codec:
-        #       h264, vn
-        # default: h264
-        hls_vcodec      h264;
-        # whether cleanup the old expired ts files.
-        # default: on
-        hls_cleanup     on;
-        # the timeout in seconds to dispose the hls,
-        # dispose is to remove all hls files, m3u8 and ts files.
-        # when publisher timeout dispose hls.
-        # @remark 0 to disable dispose for publisher.
-        # @remark apply for publisher timeout only, while "etc/init.d/srs stop" always dispose hls.
-        # default: 0
-        hls_dispose     0;
-        # the max size to notify hls,
-        # to read max bytes from ts of specified cdn network,
-        # @remark only used when on_hls_notify is config.
-        # default: 64
-        hls_nb_notify   64;
-        # whether wait keyframe to reap segment,
-        # if off, reap segment when duration exceed the fragment,
-        # if on, reap segment when duration exceed and got keyframe.
-        # default: on
-        hls_wait_keyframe       on;
-
-        # whether using AES encryption.
-        # default: off
-        hls_keys        on; 
-        # the number of clear ts which one key can encrypt.
-        # default: 5
-        hls_fragments_per_key 5;
-        # the hls key file name.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        #       [seq], the sequence number of key corresponding to the ts.
-        hls_key_file     [app]/[stream]-[seq].key;
-        # the key output path.
-        # the key file is configed by hls_path/hls_key_file, the default is:
-        # ./objs/nginx/html/[app]/[stream]-[seq].key
-        hls_key_file_path    ./objs/nginx/html;
-        # the key root URL, use this can support https.
-        # @remark It's optional.
-        hls_key_url       https://localhost:8080;
-
-        # on_hls, never config in here, should config in http_hooks.
-        # for the hls http callback, @see http_hooks.on_hls of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
-
-        # on_hls_notify, never config in here, should config in http_hooks.
-        # we support the variables to generate the notify url:
-        #       [app], replace with the app.
-        #       [stream], replace with the stream.
-        #       [param], replace with the param.
-        #       [ts_url], replace with the ts url.
-        # for the hls http callback, @see http_hooks.on_hls_notify of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#on-hls-notify
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#on-hls-notify
-    }
-}
-# the vhost with hls disabled.
-vhost no-hls.srs.com {
-    hls {
-        # whether the hls is enabled.
-        # if off, do not write hls(ts and m3u8) when publish.
-        # default: off
-        enabled         off;
-    }
-}
-
-# the vhost with adobe hds
-vhost hds.srs.com {
-    hds {
-        # whether hds enabled
-        # default: off
-        enabled         on;
-        # the hds fragment in seconds.
-        # default: 10
-        hds_fragment    10;
-        # the hds window in seconds, erase the segment when exceed the window.
-        # default: 60
-        hds_window      60;
-        # the path to store the hds files.
-        # default: ./objs/nginx/html
-        hds_path        ./objs/nginx/html;
-    }
-}
-
-# vhost for dvr
-vhost dvr.srs.com {
-    # DVR RTMP stream to file,
-    # start to record to file when encoder publish,
-    # reap flv/mp4 according by specified dvr_plan.
-    dvr {
-        # whether enabled dvr features
-        # default: off
-        enabled         on;
-        # the filter for dvr to apply to.
-        #       all, dvr all streams of all apps.
-        #       <app>/<stream>, apply to specified stream of app.
-        # for example, to dvr the following two streams:
-        #       live/stream1 live/stream2
-        # default: all
-        dvr_apply       all;
-        # the dvr plan. canbe:
-        #       session reap flv/mp4 when session end(unpublish).
-        #       segment reap flv/mp4 when flv duration exceed the specified dvr_duration.
-        # @remark The plan append is removed in SRS3+, for it's no use.
-        # default: session
-        dvr_plan        session;
-        # the dvr output path, *.flv or *.mp4.
-        # we supports some variables to generate the filename.
-        #       [vhost], the vhost of stream.
-        #       [app], the app of stream.
-        #       [stream], the stream name of stream.
-        #       [2006], replace this const to current year.
-        #       [01], replace this const to current month.
-        #       [02], replace this const to current date.
-        #       [15], replace this const to current hour.
-        #       [04], replace this const to current minute.
-        #       [05], replace this const to current second.
-        #       [999], replace this const to current millisecond.
-        #       [timestamp],replace this const to current UNIX timestamp in ms.
-        # @remark we use golang time format "2006-01-02 15:04:05.999" as "[2006]-[01]-[02]_[15].[04].[05]_[999]"
-        # for example, for url rtmp://ossrs.net/live/livestream and time 2015-01-03 10:57:30.776
-        # 1. No variables, the rule of SRS1.0(auto add [stream].[timestamp].flv as filename):
-        #       dvr_path ./objs/nginx/html;
-        #       =>
-        #       dvr_path ./objs/nginx/html/live/livestream.1420254068776.flv;
-        # 2. Use stream and date as dir name, time as filename:
-        #       dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
-        #       =>
-        #       dvr_path /data/ossrs.net/live/livestream/2015/01/03/10.57.30.776.flv;
-        # 3. Use stream and year/month as dir name, date and time as filename:
-        #       dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]-[15].[04].[05].[999].flv;
-        #       =>
-        #       dvr_path /data/ossrs.net/live/livestream/2015/01/03-10.57.30.776.flv;
-        # 4. Use vhost/app and year/month as dir name, stream/date/time as filename:
-        #       dvr_path /data/[vhost]/[app]/[2006]/[01]/[stream]-[02]-[15].[04].[05].[999].flv;
-        #       =>
-        #       dvr_path /data/ossrs.net/live/2015/01/livestream-03-10.57.30.776.flv;
-        # 5. DVR to mp4:
-        #       dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].mp4;
-        #       =>
-        #       dvr_path ./objs/nginx/html/live/livestream.1420254068776.mp4;
-        # @see https://github.com/ossrs/srs/wiki/v3_CN_DVR#custom-path
-        # @see https://github.com/ossrs/srs/wiki/v3_EN_DVR#custom-path
-        #       segment,session apply it.
-        # default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
-        dvr_path        ./objs/nginx/html/[app]/[stream].[timestamp].flv;
-        # the duration for dvr file, reap if exceed, in seconds.
-        #       segment apply it.
-        #       session,append ignore.
-        # default: 30
-        dvr_duration    30;
-        # whether wait keyframe to reap segment,
-        # if off, reap segment when duration exceed the dvr_duration,
-        # if on, reap segment when duration exceed and got keyframe.
-        #       segment apply it.
-        #       session,append ignore.
-        # default: on
-        dvr_wait_keyframe       on;
-        # about the stream monotonically increasing:
-        #   1. video timestamp is monotonically increasing,
-        #   2. audio timestamp is monotonically increasing,
-        #   3. video and audio timestamp is interleaved monotonically increasing.
-        # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
-        # however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
-        # the time jitter algorithm:
-        #   1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
-        #   2. zero, only ensure stream start at zero, ignore timestamp jitter.
-        #   3. off, disable the time jitter algorithm, like atc.
-        # apply for all dvr plan.
-        # default: full
-        time_jitter             full;
-
-        # on_dvr, never config in here, should config in http_hooks.
-        # for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
-        # @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
-        # @read https://github.com/ossrs/srs/wiki/v2_EN_DVR#http-callback
-    }
-}
-
-# vhost for ingest
-vhost ingest.srs.com {
-    # ingest file/stream/device then push to SRS over RTMP.
-    # the name/id used to identify the ingest, must be unique in global.
-    # ingest id is used in reload or http api management.
-    # @remark vhost can contains multiple ingest
-    ingest livestream {
-        # whether enabled ingest features
-        # default: off
-        enabled      on;
-        # input file/stream/device
-        # @remark only support one input.
-        input {
-            # the type of input.
-            # can be file/stream/device, that is,
-            #   file: ingest file specified by url.
-            #   stream: ingest stream specified by url.
-            #   device: not support yet.
-            # default: file
-            type    file;
-            # the url of file/stream.
-            url     ./doc/source.200kbps.768x320.flv;
-        }
-        # the ffmpeg
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        # the transcode engine, @see all.transcode.srs.com
-        # @remark, the output is specified following.
-        engine {
-            # @see enabled of transcode engine.
-            # if disabled or vcodec/acodec not specified, use copy.
-            # default: off.
-            enabled          off;
-            # output stream. variables:
-            #       [vhost] current vhost which start the ingest.
-            #       [port] system RTMP stream port.
-            output          rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
-        }
-    }
-}
-
-# the vhost for ingest with transcode engine.
-vhost transcode.ingest.srs.com {
-    ingest livestream {
-        enabled      on;
-        input {
-            type    file;
-            url     ./doc/source.200kbps.768x320.flv;
-        }
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine {
-            enabled         off;
-            perfile {
-                re;
-                rtsp_transport tcp;
-            }
-            iformat         flv;
-            vfilter {
-                i               ./doc/ffmpeg-logo.png;
-                filter_complex  'overlay=10:10';
-            }
-            vcodec          libx264;
-            vbitrate        1500;
-            vfps            25;
-            vwidth          768;
-            vheight         320;
-            vthreads        12;
-            vprofile        main;
-            vpreset         medium;
-            vparams {
-                t               100;
-                coder           1;
-                b_strategy      2;
-                bf              3;
-                refs            10;
-            }
-            acodec          libfdk_aac;
-            abitrate        70;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-                profile:a   aac_low;
-            }
-            oformat         flv;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream];
-        }
-    }
-}
-
-# the main comments for transcode
-vhost example.transcode.srs.com {
-    # the streaming transcode configs.
-    # @remark vhost can contains multiple transcode
-    transcode {
-        # whether the transcode enabled.
-        # if off, donot transcode.
-        # default: off.
-        enabled     on;
-        # the ffmpeg
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        # the transcode engine for matched stream.
-        # all matched stream will transcoded to the following stream.
-        # the transcode set name(ie. hd) is optional and not used.
-        # we will build the parameters to fork ffmpeg:
-        #       ffmpeg <perfile>
-        #               -i <iformat> 
-        #               <vfilter> 
-        #               -vcodec <vcodec> -b:v <vbitrate> -r <vfps> -s <vwidth>x<vheight> -profile:v <vprofile> -preset <vpreset>
-        #               <vparams>
-        #               -acodec <acodec> -b:a <abitrate> -ar <asample_rate> -ac <achannels>
-        #               <aparams>
-        #               -f <oformat>
-        #               -y <output>
-        engine example {
-            # whether the engine is enabled
-            # default: off.
-            enabled         on;
-            # pre-file options, before "-i"
-            perfile {
-                re;
-                rtsp_transport tcp;
-            }
-            # input format "-i", can be:
-            #       off, do not specifies the format, ffmpeg will guess it.
-            #       flv, for flv or RTMP stream.
-            #       other format, for example, mp4/aac whatever.
-            # default: flv
-            iformat         flv;
-            # ffmpeg filters, between "-i" and "-vcodec"
-            # follows the main input.
-            vfilter {
-                # the logo input file.
-                i               ./doc/ffmpeg-logo.png;
-                # the ffmpeg complex filter.
-                # for filters, @see: http://ffmpeg.org/ffmpeg-filters.html
-                filter_complex  'overlay=10:10';
-            }
-            # video encoder name, "ffmpeg -vcodec"
-            # can be:
-            #       libx264: use h.264(libx264) video encoder.
-            #       png: use png to snapshot thumbnail.
-            #       copy: donot encoder the video stream, copy it.
-            #       vn: disable video output.
-            vcodec          libx264;
-            # video bitrate, in kbps, "ffmepg -b:v"
-            # @remark 0 to use source video bitrate.
-            # default: 0
-            vbitrate        1500;
-            # video framerate, "ffmepg -r"
-            # @remark 0 to use source video fps.
-            # default: 0
-            vfps            25;
-            # video width, must be even numbers, "ffmepg -s"
-            # @remark 0 to use source video width.
-            # default: 0
-            vwidth          768;
-            # video height, must be even numbers, "ffmepg -s"
-            # @remark 0 to use source video height.
-            # default: 0
-            vheight         320;
-            # the max threads for ffmpeg to used, "ffmepg -thread"
-            # default: 1
-            vthreads        12;
-            # x264 profile, "ffmepg -profile:v"
-            # @see x264 -help, can be:
-            # high,main,baseline
-            vprofile        main;
-            # x264 preset, "ffmpeg -preset"
-            # @see x264 -help, can be:
-            #       ultrafast,superfast,veryfast,faster,fast
-            #       medium,slow,slower,veryslow,placebo
-            vpreset         medium;
-            # other x264 or ffmpeg video params, between "-preset" and "-acodec"
-            vparams {
-                # ffmpeg options, @see: http://ffmpeg.org/ffmpeg.html
-                t               100;
-                # 264 params, @see: http://ffmpeg.org/ffmpeg-codecs.html#libx264
-                coder           1;
-                b_strategy      2;
-                bf              3;
-                refs            10;
-            }
-            # audio encoder name, "ffmpeg -acodec"
-            # can be:
-            #       libfdk_aac: use aac(libfdk_aac) audio encoder.
-            #       copy: donot encoder the audio stream, copy it.
-            #       an: disable audio output.
-            acodec          libfdk_aac;
-            # audio bitrate, in kbps, "ffmpeg -b:a"
-            # [16, 72] for libfdk_aac.
-            # @remark 0 to use source audio bitrate.
-            # default: 0
-            abitrate        70;
-            # audio sample rate, "ffmpeg -ar"
-            # for flv/rtmp, it must be:
-            #       44100,22050,11025,5512
-            # @remark 0 to use source audio sample rate.
-            # default: 0
-            asample_rate    44100;
-            # audio channel, "ffmpeg -ac"
-            # 1 for mono, 2 for stereo.
-            # @remark 0 to use source audio channels.
-            # default: 0
-            achannels       2;
-            # other ffmpeg audio params, between "-ac" and "-f"/"-y"
-            aparams {
-                # audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
-                # @remark SRS supported aac profile for HLS is: aac_low, aac_he, aac_he_v2
-                profile:a   aac_low;
-                bsf:a       aac_adtstoasc;
-            }
-            # output format, "ffmpeg -f" can be:
-            #       off, do not specifies the format, ffmpeg will guess it.
-            #       flv, for flv or RTMP stream.
-            #       image2, for vcodec png to snapshot thumbnail.
-            #       other format, for example, mp4/aac whatever.
-            # default: flv
-            oformat         flv;
-            # output stream, "ffmpeg -y", variables:
-            #       [vhost] the input stream vhost.
-            #       [port] the input stream port.
-            #       [app] the input stream app.
-            #       [stream] the input stream name.
-            #       [engine] the transcode engine name.
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
-vhost mirror.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine mirror {
-            enabled         on;
-            vfilter {
-                vf                  'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
-            }
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# the drawtext filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
-# remark: we remove the libfreetype which always cause build failed, you must add it manual if needed.
-#######################################################################################################
-# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
-vhost crop.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine crop {
-            enabled         on;
-            vfilter {
-                vf                  'crop=in_w-20:in_h-160:10:80';
-            }
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# the logo filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#overlay
-vhost logo.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine logo {
-            enabled         on;
-            vfilter {
-                i               ./doc/ffmpeg-logo.png;
-                filter_complex      'overlay=10:10';
-            }
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# audio transcode only.
-# for example, FMLE publish audio codec in mp3, and do not support HLS output,
-# we can transcode the audio to aac and copy video to the new stream with HLS.
-vhost audio.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine acodec {
-            enabled         on;
-            vcodec          copy;
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# disable video, transcode/copy audio.
-# for example, publish pure audio stream.
-vhost vn.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine vn {
-            enabled         on;
-            vcodec          vn;
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# ffmpeg-copy(forward implements by ffmpeg).
-# copy the video and audio to a new stream.
-vhost copy.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine copy {
-            enabled         on;
-            vcodec          copy;
-            acodec          copy;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# transcode all app and stream of vhost
-# the comments, read example.transcode.srs.com
-vhost all.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine ffsuper {
-            enabled         on;
-            iformat         flv;
-            vfilter {
-                i               ./doc/ffmpeg-logo.png;
-                filter_complex  'overlay=10:10';
-            }
-            vcodec          libx264;
-            vbitrate        1500;
-            vfps            25;
-            vwidth          768;
-            vheight         320;
-            vthreads        12;
-            vprofile        main;
-            vpreset         medium;
-            vparams {
-                t               100;
-                coder           1;
-                b_strategy      2;
-                bf              3;
-                refs            10;
-            }
-            acodec          libfdk_aac;
-            abitrate        70;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-                profile:a   aac_low;
-            }
-            oformat         flv;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine ffhd {
-            enabled         on;
-            vcodec          libx264;
-            vbitrate        1200;
-            vfps            25;
-            vwidth          1382;
-            vheight         576;
-            vthreads        6;
-            vprofile        main;
-            vpreset         medium;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        70;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine ffsd {
-            enabled         on;
-            vcodec          libx264;
-            vbitrate        800;
-            vfps            25;
-            vwidth          1152;
-            vheight         480;
-            vthreads        4;
-            vprofile        main;
-            vpreset         fast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        60;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine fffast {
-            enabled     on;
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine vcopy {
-            enabled         on;
-            vcodec          copy;
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine acopy {
-            enabled     on;
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          copy;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-        engine copy {
-            enabled         on;
-            vcodec          copy;
-            acodec          copy;
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# transcode all stream using the empty ffmpeg demo, do nothing.
-vhost ffempty.transcode.srs.com {
-    transcode {
-        enabled     on;
-        ffmpeg ./objs/research/ffempty;
-        engine empty {
-            enabled         on;
-            vcodec          libx264;
-            vbitrate        300;
-            vfps            20;
-            vwidth          768;
-            vheight         320;
-            vthreads        2;
-            vprofile        baseline;
-            vpreset         superfast;
-            vparams {
-            }
-            acodec          libfdk_aac;
-            abitrate        45;
-            asample_rate    44100;
-            achannels       2;
-            aparams {
-            }
-            output          rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
-        }
-    }
-}
-# transcode all app and stream of app
-vhost app.transcode.srs.com {
-    # the streaming transcode configs.
-    # if app specified, transcode all streams of app.
-    transcode live {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine {
-            enabled     off;
-        }
-    }
-}
-# transcode specified stream.
-vhost stream.transcode.srs.com {
-    # the streaming transcode configs.
-    # if stream specified, transcode the matched stream.
-    transcode live/livestream {
-        enabled     on;
-        ffmpeg      ./objs/ffmpeg/bin/ffmpeg;
-        engine {
-            enabled     off;
-        }
-    }
-}
-
-#############################################################################################
-# The origin cluster section
-#############################################################################################
-http_api {
-    enabled         on;
-    listen          9090;
-}
-vhost a.origin.cluster.srs.com {
-    cluster {
-        mode                local;
-        origin_cluster      on;
-        coworkers           127.0.0.1:9091;
-    }
-}
-
-http_api {
-    enabled         on;
-    listen          9091;
-}
-vhost b.origin.cluster.srs.com {
-    cluster {
-        mode                local;
-        origin_cluster      on;
-        coworkers           127.0.0.1:9090;
-    }
-}
diff --git a/publish/src/linux-x64/publish/apps/srs/conf/srs.conf b/publish/src/linux-x64/publish/apps/srs/conf/srs.conf
index bcff0ae3..3fcbb7df 100644
--- a/publish/src/linux-x64/publish/apps/srs/conf/srs.conf
+++ b/publish/src/linux-x64/publish/apps/srs/conf/srs.conf
@@ -38,7 +38,7 @@ vhost __defaultVhost__ {
     dvr {
         enabled on;
         dvr_apply none;
-        dvr_path ../IoTCenter/wwwroot/video/[app]-[stream]-[timestamp].mp4;
+        dvr_path ./objs/nginx/html/[app].[stream].[timestamp].flv;;
         dvr_plan session;
         dvr_duration 300;
         dvr_wait_keyframe on;
@@ -46,6 +46,6 @@ vhost __defaultVhost__ {
     }
     http_hooks {
         enabled on;
-        on_dvr http://localhost:8011/App/OnDvr;
+        on_dvr http://localhost/IoTCenter/App/OnDvr;
     }
 }