A signaling server for GB28181

.gitignore

@@ -21,3 +21,6 @@
 
 # virtual machine crash logs, see http://www.java.com/en/download/help/error_hotspot.xml
 hs_err_pid*
+bin/
+objs
+.idea

.vscode/launch.json

@@ -0,0 +1,17 @@
+{
+    // 使用 IntelliSense 了解相关属性。 
+    // 悬停以查看现有属性的描述。
+    // 欲了解更多信息，请访问: https://go.microsoft.com/fwlink/?linkid=830387
+    "version": "0.2.0",
+    "configurations": [
+        {
+            "name": "Launch Package",
+            "type": "go",
+            "request": "launch",
+            "mode": "auto",
+            "program": "${workspaceFolder}/main",
+            "env": {},
+            "args": ["-sip-port", "5080", "-media-addr", "127.0.0.1:1985"]
+        }
+    ]
+}

LICENSE

@@ -1,6 +1,6 @@
 MIT License
 
-Copyright (c) 2022 ossrs
+Copyright (c) 2024 Haibo Chen
 
 Permission is hereby granted, free of charge, to any person obtaining a copy
 of this software and associated documentation files (the "Software"), to deal

Makefile

@@ -0,0 +1,22 @@
+GOCMD=go
+GOBUILD=$(GOCMD) build
+BINARY_NAME=bin/srs-sip
+MAIN_PATH=main/main.go
+
+default: build
+
+build:
+	$(GOBUILD) -o $(BINARY_NAME) $(MAIN_PATH)
+
+clean:
+	rm -f $(BINARY_NAME)
+
+run:
+	$(GOBUILD) -o $(BINARY_NAME) $(MAIN_PATH)
+	./$(BINARY_NAME)
+
+install:
+	$(GOBUILD) -o $(BINARY_NAME) $(MAIN_PATH)
+	mv $(BINARY_NAME) /usr/local/bin
+
+.PHONY: clean

README.md

@@ -0,0 +1,74 @@
+# SRS-SIP
+
+## Usage
+
+Pre-requisites:
+- Go 1.20+ is installed
+- GOPATH/bin is in your PATH
+
+Then run
+```
+git clone https://github.com/ossrs/srs-sip
+cd srs-sip
+./bootstrap.sh
+mage
+```
+
+If you are on a Unix-like system, you can also run the following command.
+```
+make
+```
+
+Run the program:
+
+```
+./bin/srs-sip -sip-port 5060 -media-addr 127.0.0.1:1985 -api-port 2020 -http-server-port 8888
+```
+
+- `sip-port` : the SIP port, this program listen on, for device register with gb28181
+- `media-addr` : the API address for SRS, typically on port 1985, used to send HTTP requests to "/gb/v1/publish"
+- `api-port`: The API server port, used to send HTTP requests, for example "/srs-sip/v1/channels"
+- `http-server-port`: The demo web server.
+
+Access http://localhost:8888 in web browser.
+
+## Sequence
+
+1. 注册流程
+```mermaid
+sequenceDiagram
+    Device ->> SRS-SIP : 1. Register
+    SRS-SIP ->> Device : 2. 200 OK
+```
+
+暂时没有实现鉴权功能，敬请期待。
+
+2. 播放视频流程
+Player、SRS-SIP、SRS Server和GB28181 Device的交互图如下：
+
+```mermaid
+sequenceDiagram
+    Player ->> SRS-SIP : 1. Play Request(with id)
+    SRS-SIP ->>  SRS : 2. Publish Request(with ssrc and id)
+    SRS ->> SRS-SIP : 3. Response(with port)
+    SRS-SIP ->>  Device : 4. Invite(with port)
+    Device ->> SRS-SIP : 5. 200 OK
+    SRS-SIP ->> Player : 6. 200 OK(with url)
+    Device -->> SRS : Media Stream
+    Player ->> SRS : 7. Play
+    SRS -->> Player : Media Stream
+    Player ->> SRS-SIP : 8. Stop Request
+    SRS-SIP ->> SRS : 9. Unpublish Request
+    SRS-SIP ->> Device : 10. Bye
+```
+
+1. 通过SRS-SIP提供的API接口`/srs-sip/v1/invite`，Player主动发起播放请求，携带设备的通道ID
+2. SRS-SIP向SRS发起推流请求，携带SSRC和ID，SSRC是设备推流时RTP里的字段
+3. SRS响应推流请求，并返回收流端口。目前SRS仅支持TCP单端口模式，在配置文件`stream_caster.listen`中配置
+4. SRS-SIP通过GB28181协议向设备发起`Invite`请求，携带SRS的收流端口及SSRC
+5. 设备响应成功
+6. SRS-SIP响应成功，携带URL，用于播放
+7. Player通过返回的URL进行拉流播放
+8. Player停止播放
+9. SRS-SIP通知SRS停止收流
+10. SRS-SIP通过设备停止推流

bootstrap.sh

@@ -0,0 +1,49 @@
+add_gopath_to_path() {
+    GOPATH=$(go env GOPATH)
+    # Check if GOPATH is set
+    if [ -z "$GOPATH" ]; then
+        echo "GOPATH is not set."
+        return 1
+    fi
+
+    # Check if $GOPATH/bin is already in ~/.bashrc
+    if grep -q "$GOPATH/bin" ~/.bashrc; then
+        echo "$GOPATH/bin is already in PATH."
+        return 0
+    fi
+
+    # Add $GOPATH/bin to PATH
+    echo "export PATH=\$PATH:$GOPATH/bin" >> ~/.bashrc
+    source ~/.bashrc
+
+    echo "$GOPATH/bin has been added to PATH."
+}
+
+
+if ! command -v mage &> /dev/null
+then
+  pushd /tmp
+
+  OS_IS_LINUX=$(uname -s |grep -q Linux && echo YES)
+  if [ "$OS_IS_LINUX" == "YES" ]; then
+    add_gopath_to_path
+    if [ $? -eq 1 ]; then
+        echo "error: Failed to add $GOPATH/bin to PATH."
+        exit 1
+    fi
+  fi
+
+  git clone https://github.com/magefile/mage
+  cd mage
+  go run bootstrap.go
+  rm -rf /tmp/mage
+  popd
+fi
+
+if ! command -v mage &> /dev/null
+then
+  echo "error: Ensure `go env GOPATH`/bin is in your \$PATH"
+  exit 1
+fi
+
+go mod download

go.mod

@@ -0,0 +1,41 @@
+module github.com/ossrs/srs-sip
+
+go 1.20
+
+require (
+	github.com/gorilla/handlers v1.5.2
+	github.com/gorilla/mux v1.8.1
+	github.com/magefile/mage v1.15.0
+	github.com/ossrs/go-oryx-lib v0.0.9
+	github.com/ossrs/srs-bench v0.0.0-20230906232735-aa029b492d0f
+	github.com/rs/zerolog v1.32.0
+	golang.org/x/net v0.10.0
+)
+
+require (
+	github.com/emiago/sipgo v0.22.1 // indirect
+	github.com/felixge/httpsnoop v1.0.3 // indirect
+	github.com/ghettovoice/gosip v0.0.0-20220929080231-de8ba881be83 // indirect
+	github.com/gobwas/httphead v0.1.0 // indirect
+	github.com/gobwas/pool v0.2.1 // indirect
+	github.com/gobwas/ws v1.3.2 // indirect
+	github.com/google/uuid v1.6.0 // indirect
+	github.com/icholy/digest v0.1.22 // indirect
+	github.com/kr/text v0.2.0 // indirect
+	github.com/mattn/go-colorable v0.1.13 // indirect
+	github.com/mattn/go-isatty v0.0.20 // indirect
+	github.com/mgutz/ansi v0.0.0-20170206155736-9520e82c474b // indirect
+	github.com/pion/randutil v0.1.0 // indirect
+	github.com/pion/rtp v1.7.13 // indirect
+	github.com/pion/webrtc/v3 v3.2.9 // indirect
+	github.com/satori/go.uuid v1.2.1-0.20181028125025-b2ce2384e17b // indirect
+	github.com/sirupsen/logrus v1.9.3 // indirect
+	github.com/tevino/abool v0.0.0-20170917061928-9b9efcf221b5 // indirect
+	github.com/x-cray/logrus-prefixed-formatter v0.5.2 // indirect
+	github.com/yapingcat/gomedia/codec v0.0.0-20220617074658-94762898dc25 // indirect
+	github.com/yapingcat/gomedia/mpeg2 v0.0.0-20220617074658-94762898dc25 // indirect
+	golang.org/x/crypto v0.9.0 // indirect
+	golang.org/x/sys v0.19.0 // indirect
+	golang.org/x/term v0.8.0 // indirect
+	golang.org/x/text v0.9.0 // indirect
+)

go.sum

@@ -0,0 +1,271 @@
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magefile.go

@@ -0,0 +1,50 @@
+//go:build mage
+// +build mage
+
+package main
+
+import (
+	"fmt"
+	"os"
+	"path/filepath"
+	"runtime"
+
+	"github.com/magefile/mage/sh"
+)
+
+var Default = Build
+
+func Build() error {
+	path := "bin"
+	if err := os.MkdirAll(path, 0755); err != nil {
+		return err
+	}
+
+	name := "srs-sip"
+	if runtime.GOOS == "windows" {
+		name += ".exe"
+	}
+	name = filepath.Join(path, name)
+
+	if err := sh.Run("go", "build", "-o", name, "main/main.go"); err != nil {
+		return err
+	}
+
+	name = "srs-sip-tools"
+	if runtime.GOOS == "windows" {
+		name += ".exe"
+	}
+	name = filepath.Join(path, name)
+
+	if err := sh.Run("go", "build", "-o", name, "tools/main.go"); err != nil {
+		return err
+	}
+
+	fmt.Println("build done")
+	return nil
+}
+
+func Clean() {
+	os.RemoveAll("bin")
+	fmt.Println("clean done")
+}

main/main.go

@@ -0,0 +1,85 @@
+package main
+
+import (
+	"context"
+	"net/http"
+	"os"
+	"os/signal"
+	"path"
+	"strconv"
+	"syscall"
+	"time"
+
+	"github.com/ossrs/go-oryx-lib/logger"
+	"github.com/ossrs/srs-sip/pkg/api"
+	"github.com/ossrs/srs-sip/pkg/config"
+	"github.com/ossrs/srs-sip/pkg/service"
+	"github.com/ossrs/srs-sip/pkg/utils"
+)
+
+func WaitTerminationSignal(cancel context.CancelFunc) {
+	sigc := make(chan os.Signal, 1)
+	signal.Notify(sigc, syscall.SIGINT, syscall.SIGTERM)
+	defer signal.Stop(sigc)
+	<-sigc
+	cancel()
+}
+
+func main() {
+	ctx, cancel := context.WithCancel(context.Background())
+
+	conf := utils.Parse(ctx)
+	sipSvr, err := service.NewService(ctx, conf)
+	if err != nil {
+		logger.Ef("create service failed. err is %v", err.Error())
+		return
+	}
+
+	if err := sipSvr.Start(); err != nil {
+		logger.Ef("start sip service failed. err is %v", err.Error())
+		return
+	}
+
+	apiSvr, err := api.NewHttpApiServer(conf, sipSvr)
+	if err != nil {
+		logger.Ef("create http service failed. err is %v", err.Error())
+		return
+	}
+	apiSvr.Start()
+
+	var targetDir string
+	targetDirs := []string{"./web/html", "../web/html"}
+	for _, dir := range targetDirs {
+		if _, err := os.Stat(path.Join(dir, "index.html")); err == nil {
+			targetDir = dir
+			break
+		}
+	}
+	if targetDir == "" {
+		logger.Ef(ctx, "index.html not found in %v", targetDirs)
+		return
+	}
+
+	go func() {
+		c := conf.(*config.MainConfig)
+		httpPort := strconv.Itoa(c.HttpServerPort)
+		server := &http.Server{
+			Addr:              ":" + httpPort,
+			Handler:           http.FileServer(http.Dir(targetDir)),
+			ReadTimeout:       10 * time.Second,
+			WriteTimeout:      10 * time.Second,
+			IdleTimeout:       30 * time.Second,
+			ReadHeaderTimeout: 5 * time.Second,
+		}
+		logger.Tf(ctx, "http server listen on %s, home is %v", httpPort, targetDir)
+		if err := server.ListenAndServe(); err != nil && err != http.ErrServerClosed {
+			logger.Ef(ctx, "listen on %s failed", httpPort)
+		}
+	}()
+
+	logger.Tf(ctx, "media server address is %v", conf.(*config.MainConfig).MediaAddr)
+
+	WaitTerminationSignal(cancel)
+
+	sipSvr.Stop()
+}

pkg/api/api-controller.go

@@ -0,0 +1,137 @@
+package api
+
+import (
+	"encoding/json"
+	"net/http"
+
+	"github.com/gorilla/mux"
+	"github.com/ossrs/srs-sip/pkg/service"
+)
+
+func (h *HttpApiServer) RegisterRoutes(router *mux.Router) {
+
+	apiV1Router := router.PathPrefix("/srs-sip/v1").Subrouter()
+
+	// Add Auth middleware
+	//apiV1Router.Use(authMiddleware)
+
+	apiV1Router.HandleFunc("/devices", h.ApiListDevices).Methods(http.MethodGet)
+	apiV1Router.HandleFunc("/devices/{id}/channels", h.ApiGetChannelByDeviceId).Methods(http.MethodGet)
+	apiV1Router.HandleFunc("/channels", h.ApiGetAllChannels).Methods(http.MethodGet)
+
+	apiV1Router.HandleFunc("/invite", h.ApiInvite).Methods(http.MethodPost)
+	apiV1Router.HandleFunc("/bye", h.ApiBye).Methods(http.MethodPost)
+	apiV1Router.HandleFunc("/ptz", h.ApiPTZControl).Methods(http.MethodPost)
+
+	apiV1Router.HandleFunc("", h.GetAPIRoutes(apiV1Router)).Methods(http.MethodGet)
+
+	router.HandleFunc("/srs-sip", h.ApiGetAPIVersion).Methods(http.MethodGet)
+}
+
+func (h *HttpApiServer) RespondWithJSON(w http.ResponseWriter, code int, data interface{}) {
+	w.Header().Set("Content-Type", "application/json")
+	wrapper := map[string]interface{}{
+		"code": code,
+		"data": data,
+	}
+	json.NewEncoder(w).Encode(wrapper)
+}
+
+func (h *HttpApiServer) RespondWithJSONSimple(w http.ResponseWriter, jsonStr string) {
+	w.Header().Set("Content-Type", "application/json")
+	w.Write([]byte(jsonStr))
+}
+
+func (h *HttpApiServer) GetAPIRoutes(router *mux.Router) http.HandlerFunc {
+	return func(w http.ResponseWriter, r *http.Request) {
+		var routes []map[string]string
+
+		router.Walk(func(route *mux.Route, router *mux.Router, ancestors []*mux.Route) error {
+			path, err := route.GetPathTemplate()
+			if err != nil {
+				return err
+			}
+			methods, err := route.GetMethods()
+			if err != nil {
+				return err
+			}
+			for _, method := range methods {
+				routes = append(routes, map[string]string{
+					"method": method,
+					"path":   path,
+				})
+			}
+			return nil
+		})
+
+		h.RespondWithJSON(w, 0, routes)
+	}
+}
+
+func (h *HttpApiServer) ApiGetAPIVersion(w http.ResponseWriter, r *http.Request) {
+	h.RespondWithJSONSimple(w, `{"version": "v1"}`)
+}
+
+func (h *HttpApiServer) ApiListDevices(w http.ResponseWriter, r *http.Request) {
+	list := service.DM.GetDevices()
+	h.RespondWithJSON(w, 0, list)
+}
+
+func (h *HttpApiServer) ApiGetChannelByDeviceId(w http.ResponseWriter, r *http.Request) {
+	vars := mux.Vars(r)
+	id := vars["id"]
+	channels := service.DM.ApiGetChannelByDeviceId(id)
+	h.RespondWithJSON(w, 0, channels)
+}
+
+func (h *HttpApiServer) ApiGetAllChannels(w http.ResponseWriter, r *http.Request) {
+	channels := service.DM.GetAllVideoChannels()
+	h.RespondWithJSON(w, 0, channels)
+}
+
+// request: {"device_id": "1", "channel_id": "1", "sub_stream": 0}
+// response: {"code": 0, "data": {"channel_id": "1", "url": "webrtc://"}}
+func (h *HttpApiServer) ApiInvite(w http.ResponseWriter, r *http.Request) {
+	// Parse request
+	var req map[string]string
+	err := json.NewDecoder(r.Body).Decode(&req)
+	if err != nil {
+		w.WriteHeader(http.StatusBadRequest)
+		return
+	}
+
+	// Get device and channel
+	deviceID := req["device_id"]
+	channelID := req["channel_id"]
+	//subStream := req["sub_stream"]
+
+	code := 0
+	url := ""
+
+	defer func() {
+		data := map[string]string{
+			"channel_id": channelID,
+			"url":        url,
+		}
+		h.RespondWithJSON(w, code, data)
+	}()
+
+	if err := h.sipSvr.Uas.Invite(deviceID, channelID); err != nil {
+		code = http.StatusInternalServerError
+		return
+	}
+	c, ok := h.sipSvr.Uas.GetVideoChannelStatue(channelID)
+	if !ok {
+		code = http.StatusInternalServerError
+		return
+	}
+	url = "webrtc://" + h.conf.MediaAddr + "/live/" + c.Ssrc
+}
+
+func (h *HttpApiServer) ApiBye(w http.ResponseWriter, r *http.Request) {
+	h.RespondWithJSONSimple(w, `{"msg":"Not implemented"}`)
+}
+
+func (h *HttpApiServer) ApiPTZControl(w http.ResponseWriter, r *http.Request) {
+	h.RespondWithJSONSimple(w, `{"msg":"Not implemented"}`)
+}

pkg/api/api.go

@@ -0,0 +1,45 @@
+package api
+
+import (
+	"context"
+	"fmt"
+	"net/http"
+
+	"github.com/ossrs/go-oryx-lib/logger"
+
+	"github.com/gorilla/handlers"
+	"github.com/gorilla/mux"
+	"github.com/ossrs/srs-sip/pkg/config"
+	"github.com/ossrs/srs-sip/pkg/service"
+)
+
+type HttpApiServer struct {
+	conf   *config.MainConfig
+	sipSvr *service.Service
+}
+
+func NewHttpApiServer(r0 interface{}, svr *service.Service) (*HttpApiServer, error) {
+	return &HttpApiServer{
+		conf:   r0.(*config.MainConfig),
+		sipSvr: svr,
+	}, nil
+}
+
+func (h *HttpApiServer) Start() {
+	router := mux.NewRouter().StrictSlash(true)
+	h.RegisterRoutes(router)
+
+	headers := handlers.AllowedHeaders([]string{"X-Requested-With", "Content-Type", "Authorization"})
+	methods := handlers.AllowedMethods([]string{"GET", "POST", "PUT", "DELETE", "OPTIONS"})
+	origins := handlers.AllowedOrigins([]string{"*"})
+
+	go func() {
+		ctx := context.Background()
+		addr := fmt.Sprintf(":%v", h.conf.APIPort)
+		logger.Tf(ctx, "http api listen on %s", addr)
+		err := http.ListenAndServe(addr, handlers.CORS(headers, methods, origins)(router))
+		if err != nil {
+			panic(err)
+		}
+	}()
+}

pkg/config/config.go

@@ -0,0 +1,56 @@
+package config
+
+import (
+	"fmt"
+	"net"
+)
+
+type MainConfig struct {
+	Serial         string `ymal:"serial"`
+	Realm          string `ymal:"realm"`
+	SipHost        string `ymal:"sip-host"`
+	SipPort        int    `ymal:"sip-port"`
+	MediaAddr      string `ymal:"media-addr"`
+	HttpServerPort int    `ymal:"http-server-port"`
+	APIPort        int    `ymal:"api-port"`
+}
+
+func GetLocalIP() (string, error) {
+	ifaces, err := net.Interfaces()
+	if err != nil {
+		return "", nil
+	}
+	type Iface struct {
+		Name string
+		Addr net.IP
+	}
+	var candidates []Iface
+	for _, ifc := range ifaces {
+		if ifc.Flags&net.FlagUp == 0 || ifc.Flags&net.FlagUp == 0 {
+			continue
+		}
+		if ifc.Flags&(net.FlagPointToPoint|net.FlagLoopback) != 0 {
+			continue
+		}
+		addrs, err := ifc.Addrs()
+		if err != nil {
+			continue
+		}
+		for _, addr := range addrs {
+			ipnet, ok := addr.(*net.IPNet)
+			if !ok {
+				continue
+			}
+			if ip4 := ipnet.IP.To4(); ip4 != nil {
+				candidates = append(candidates, Iface{
+					Name: ifc.Name, Addr: ip4,
+				})
+				//logger.Tf("considering interface", "iface", ifc.Name, "ip", ip4)
+			}
+		}
+	}
+	if len(candidates) == 0 {
+		return "", fmt.Errorf("No local IP found")
+	}
+	return candidates[0].Addr.String(), nil
+}

pkg/service/cascade.go

@@ -0,0 +1,17 @@
+package service
+
+import (
+	"context"
+
+	"github.com/emiago/sipgo"
+	"github.com/ossrs/srs-sip/pkg/config"
+)
+
+type Cascade struct {
+	ua     *sipgo.UserAgent
+	sipCli *sipgo.Client
+	sipSvr *sipgo.Server
+
+	ctx  context.Context
+	conf *config.MainConfig
+}

pkg/service/device.go

@@ -0,0 +1,166 @@
+package service
+
+import (
+	"sync"
+
+	"github.com/ossrs/srs-sip/pkg/utils"
+)
+
+// <Item>
+// 	<DeviceID>34020000001320000002</DeviceID>
+// 	<Name>209</Name>
+// 	<Manufacturer>UNIVIEW</Manufacturer>
+// 	<Model>HIC6622-IR@X33-VF</Model>
+// 	<Owner>IPC-B2202.7.11.230222</Owner>
+// 	<CivilCode>CivilCode</CivilCode>
+// 	<Address>Address</Address>
+// 	<Parental>1</Parental>
+// 	<ParentID>75015310072008100002</ParentID>
+// 	<SafetyWay>0</SafetyWay>
+// 	<RegisterWay>1</RegisterWay>
+// 	<Secrecy>0</Secrecy>
+// 	<Status>ON</Status>
+// 	<Longitude>0.0000000</Longitude>
+// 	<Latitude>0.0000000</Latitude>
+// 	<Info>
+// 		<PTZType>1</PTZType>
+// 		<Resolution>6/4/2</Resolution>
+// 		<DownloadSpeed>0</DownloadSpeed>
+// 	</Info>
+// </Item>
+
+type ChannelInfo struct {
+	DeviceID     string        `json:"device_id"`
+	ParentID     string        `json:"parent_id"`
+	Name         string        `json:"name"`
+	Manufacturer string        `json:"manufacturer"`
+	Model        string        `json:"model"`
+	Owner        string        `json:"owner"`
+	CivilCode    string        `json:"civil_code"`
+	Address      string        `json:"address"`
+	Port         int           `json:"port"`
+	Parental     int           `json:"parental"`
+	SafetyWay    int           `json:"safety_way"`
+	RegisterWay  int           `json:"register_way"`
+	Secrecy      int           `json:"secrecy"`
+	IPAddress    string        `json:"ip_address"`
+	Status       ChannelStatus `json:"status"`
+	Longitude    float64       `json:"longitude"`
+	Latitude     float64       `json:"latitude"`
+	Info         struct {
+		PTZType       int    `json:"ptz_type"`
+		Resolution    string `json:"resolution"`
+		DownloadSpeed string `json:"download_speed"` // 1/2/4/8
+	} `json:"info"`
+
+	// custom fields
+	Ssrc string `json:"ssrc"`
+}
+
+type ChannelStatus string
+
+type DeviceInfo struct {
+	DeviceID    string   `json:"device_id"`
+	SourceAddr  string   `json:"source_addr"`
+	NetworkType string   `json:"network_type"`
+	ChannelMap  sync.Map `json:"-"`
+}
+
+type deviceManager struct {
+	devices sync.Map
+}
+
+var instance *deviceManager
+var once sync.Once
+
+func GetDeviceManager() *deviceManager {
+	once.Do(func() {
+		instance = &deviceManager{
+			devices: sync.Map{},
+		}
+	})
+	return instance
+}
+
+func (dm *deviceManager) AddDevice(id string, info *DeviceInfo) {
+	dm.devices.Store(id, info)
+}
+
+func (dm *deviceManager) RemoveDevice(id string) {
+	dm.devices.Delete(id)
+}
+
+func (dm *deviceManager) GetDevices() []*DeviceInfo {
+	list := make([]*DeviceInfo, 0)
+	dm.devices.Range(func(key, value interface{}) bool {
+		list = append(list, value.(*DeviceInfo))
+		return true
+	})
+	return list
+}
+
+func (dm *deviceManager) GetDevice(id string) (*DeviceInfo, bool) {
+	v, ok := dm.devices.Load(id)
+	if !ok {
+		return nil, false
+	}
+	return v.(*DeviceInfo), true
+}
+
+func (dm *deviceManager) UpdateChannels(deviceID string, list ...ChannelInfo) {
+	device, ok := dm.GetDevice(deviceID)
+	if !ok {
+		return
+	}
+
+	for _, channel := range list {
+		device.ChannelMap.Store(channel.DeviceID, channel)
+	}
+	dm.devices.Store(deviceID, device)
+}
+
+func (dm *deviceManager) ApiGetChannelByDeviceId(deviceID string) []ChannelInfo {
+	device, ok := dm.GetDevice(deviceID)
+	if !ok {
+		return nil
+	}
+
+	channels := make([]ChannelInfo, 0)
+	device.ChannelMap.Range(func(key, value interface{}) bool {
+		channels = append(channels, value.(ChannelInfo))
+		return true
+	})
+	return channels
+}
+
+func (dm *deviceManager) GetAllVideoChannels() []ChannelInfo {
+	channels := make([]ChannelInfo, 0)
+	dm.devices.Range(func(key, value interface{}) bool {
+		device := value.(*DeviceInfo)
+		device.ChannelMap.Range(func(key, value interface{}) bool {
+			if utils.IsVideoChannel(value.(ChannelInfo).DeviceID) {
+				channels = append(channels, value.(ChannelInfo))
+				return true
+			}
+			return true
+		})
+		return true
+	})
+	return channels
+}
+
+func (dm *deviceManager) GetDeviceInfoByChannel(channelID string) (*DeviceInfo, bool) {
+	var device *DeviceInfo
+	found := false
+	dm.devices.Range(func(key, value interface{}) bool {
+		d := value.(*DeviceInfo)
+		_, ok := d.ChannelMap.Load(channelID)
+		if ok {
+			device = d
+			found = true
+			return false
+		}
+		return true
+	})
+	return device, found
+}

pkg/service/inbound.go

@@ -0,0 +1,153 @@
+package service
+
+import (
+	"bytes"
+	"encoding/xml"
+	"net/http"
+	"strconv"
+
+	"github.com/emiago/sipgo/sip"
+	"github.com/ossrs/go-oryx-lib/logger"
+	"github.com/ossrs/srs-sip/pkg/service/stack"
+	"golang.org/x/net/html/charset"
+)
+
+const GB28181_ID_LENGTH = 20
+
+type VideoChannelStatus struct {
+	ID        string
+	ParentID  string
+	MediaHost string
+	MediaPort int
+	Ssrc      string
+	Status    string
+}
+
+func (s *UAS) onRegister(req *sip.Request, tx sip.ServerTransaction) {
+	id := req.From().Address.User
+	if len(id) != GB28181_ID_LENGTH {
+		logger.E(s.ctx, "invalid device ID")
+		return
+	}
+
+	isUnregister := false
+	if exps := req.GetHeaders("Expires"); len(exps) > 0 {
+		exp := exps[0]
+		expSec, err := strconv.ParseInt(exp.Value(), 10, 32)
+		if err != nil {
+			logger.Ef(s.ctx, "parse expires header error: %s", err.Error())
+			return
+		}
+		if expSec == 0 {
+			isUnregister = true
+		}
+	} else {
+		logger.E(s.ctx, "empty expires header")
+		return
+	}
+
+	if isUnregister {
+		DM.RemoveDevice(id)
+		logger.Wf(s.ctx, "Device %s unregistered", id)
+		return
+	} else {
+		if d, ok := DM.GetDevice(id); !ok {
+			DM.AddDevice(id, &DeviceInfo{
+				DeviceID:    id,
+				SourceAddr:  req.Source(),
+				NetworkType: req.Transport(),
+			})
+			s.respondRegister(req, http.StatusOK, "OK", tx)
+			logger.Tf(s.ctx, "%s Register success, source:%s, req: %s", id, req.Source(), req.String())
+
+			go s.Catalog(id)
+		} else {
+			if d.SourceAddr != req.Source() {
+				logger.Ef(s.ctx, "Device %s[%s] already registered, %s is NOT allowed.", id, d.SourceAddr, req.Source())
+				// TODO: 国标没有明确定义重复ID注册的处理方式，这里暂时返回冲突
+				s.respondRegister(req, http.StatusConflict, "Conflict Device ID", tx)
+			} else {
+				// TODO: 刷新DM里面的设备信息
+				s.respondRegister(req, http.StatusOK, "OK", tx)
+			}
+		}
+	}
+}
+
+func (s *UAS) respondRegister(req *sip.Request, code sip.StatusCode, reason string, tx sip.ServerTransaction) {
+	res := stack.NewRegisterResponse(req, code, reason)
+	_ = tx.Respond(res)
+
+}
+
+func (s *UAS) onMessage(req *sip.Request, tx sip.ServerTransaction) {
+	id := req.From().Address.User
+	if len(id) != 20 {
+		logger.Ef(s.ctx, "invalid device ID %s", req.String())
+	}
+
+	//logger.Tf(s.ctx, "Received MESSAGE: %s", req.String())
+
+	temp := &struct {
+		XMLName      xml.Name
+		CmdType      string
+		SN           int // 请求序列号，一般用于对应 request 和 response
+		DeviceID     string
+		DeviceName   string
+		Manufacturer string
+		Model        string
+		Channel      string
+		DeviceList   []ChannelInfo `xml:"DeviceList>Item"`
+		// RecordList   []*Record     `xml:"RecordList>Item"`
+		// SumNum       int
+	}{}
+	decoder := xml.NewDecoder(bytes.NewReader([]byte(req.Body())))
+	decoder.CharsetReader = charset.NewReaderLabel
+	if err := decoder.Decode(temp); err != nil {
+		logger.Ef(s.ctx, "decode message error: %s\n message:%s", err.Error(), req.Body())
+	}
+	var body string
+	switch temp.CmdType {
+	case "Keepalive":
+		logger.T(s.ctx, "Keepalive")
+		if _, ok := DM.GetDevice(temp.DeviceID); !ok {
+			// unregister device
+			tx.Respond(sip.NewResponseFromRequest(req, http.StatusBadRequest, "", nil))
+			return
+		}
+	case "SensorCatalog": // 兼容宇视，非国标
+	case "Catalog":
+		logger.T(s.ctx, "Catalog")
+		DM.UpdateChannels(temp.DeviceID, temp.DeviceList...)
+		//go s.AutoInvite(temp.DeviceID, temp.DeviceList...)
+	case "Alarm":
+		logger.T(s.ctx, "Alarm")
+	default:
+		logger.Wf(s.ctx, "Not supported CmdType: %s", temp.CmdType)
+		response := sip.NewResponseFromRequest(req, http.StatusBadRequest, "", nil)
+		tx.Respond(response)
+		return
+	}
+	tx.Respond(sip.NewResponseFromRequest(req, http.StatusOK, "OK", []byte(body)))
+}
+
+func (s *UAS) onNotify(req *sip.Request, tx sip.ServerTransaction) {
+	logger.T(s.ctx, "Received NOTIFY request")
+	tx.Respond(sip.NewResponseFromRequest(req, http.StatusOK, "OK", nil))
+}
+
+func (s *UAS) AddVideoChannelStatue(channelID string, status VideoChannelStatus) {
+	s.channelsStatue.Store(channelID, status)
+}
+
+func (s *UAS) GetVideoChannelStatue(channelID string) (VideoChannelStatus, bool) {
+	v, ok := s.channelsStatue.Load(channelID)
+	if !ok {
+		return VideoChannelStatus{}, false
+	}
+	return v.(VideoChannelStatus), true
+}
+
+func (s *UAS) RemoveVideoChannelStatue(channelID string) {
+	s.channelsStatue.Delete(channelID)
+}

pkg/service/outbound.go

@@ -0,0 +1,169 @@
+package service
+
+import (
+	"fmt"
+	"strings"
+
+	"github.com/emiago/sipgo/sip"
+	"github.com/ossrs/go-oryx-lib/errors"
+	"github.com/ossrs/go-oryx-lib/logger"
+	"github.com/ossrs/srs-sip/pkg/service/stack"
+	"github.com/ossrs/srs-sip/pkg/utils"
+)
+
+func (s *UAS) AutoInvite(deviceID string, list ...ChannelInfo) {
+	for _, c := range list {
+		if c.Status == "ON" && utils.IsVideoChannel(c.DeviceID) {
+			if err := s.Invite(deviceID, c.DeviceID); err != nil {
+				logger.Ef(s.ctx, "invite error: %s", err.Error())
+			}
+		}
+	}
+}
+
+func (s *UAS) Invite(deviceID, channelID string) error {
+	if s.isPublishing(channelID) {
+		return nil
+	}
+
+	ssrc := utils.CreateSSRC(true)
+
+	mediaPort, err := s.signal.Publish(ssrc, ssrc)
+	if err != nil {
+		return errors.Wrapf(err, "api gb publish request error")
+	}
+
+	mediaHost := strings.Split(s.conf.MediaAddr, ":")[0]
+	if mediaHost == "" {
+		return errors.Errorf("media host is empty")
+	}
+
+	sdpInfo := []string{
+		"v=0",
+		fmt.Sprintf("o=%s 0 0 IN IP4 %s", channelID, mediaHost),
+		"s=" + "Play",
+		"u=" + channelID + ":0",
+		"c=IN IP4 " + mediaHost,
+		"t=0 0", // start time and end time
+		fmt.Sprintf("m=video %d TCP/RTP/AVP 96", mediaPort),
+		"a=recvonly",
+		"a=rtpmap:96 PS/90000",
+		"y=" + ssrc,
+		"\r\n",
+	}
+	if true { // support tcp only
+		sdpInfo = append(sdpInfo, "a=setup:passive", "a=connection:new")
+	}
+
+	// TODO: 需要考虑不同设备，通道ID相同的情况
+	d, ok := DM.GetDeviceInfoByChannel(channelID)
+	if !ok {
+		return errors.Errorf("device not found by %s", channelID)
+	}
+
+	subject := fmt.Sprintf("%s:%s,%s:0", channelID, ssrc, s.conf.Serial)
+
+	req, err := stack.NewInviteRequest([]byte(strings.Join(sdpInfo, "\r\n")), subject, stack.OutboundConfig{
+		Via:       d.SourceAddr,
+		To:        d.DeviceID,
+		From:      s.conf.Serial,
+		Transport: d.NetworkType,
+	})
+	if err != nil {
+		return errors.Wrapf(err, "build invite request error")
+	}
+	tx, err := s.sipCli.TransactionRequest(s.ctx, req)
+	if err != nil {
+		return errors.Wrapf(err, "transaction request error")
+	}
+
+	res, err := s.waitAnswer(tx)
+	if err != nil {
+		return errors.Wrapf(err, "wait answer error")
+	}
+	if res.StatusCode != 200 {
+		return errors.Errorf("invite response error: %s", res.String())
+	}
+
+	ack := sip.NewAckRequest(req, res, nil)
+	s.sipCli.WriteRequest(ack)
+
+	s.AddVideoChannelStatue(channelID, VideoChannelStatus{
+		ID:        channelID,
+		ParentID:  deviceID,
+		MediaHost: mediaHost,
+		MediaPort: mediaPort,
+		Ssrc:      ssrc,
+		Status:    "ON",
+	})
+
+	return nil
+}
+
+func (s *UAS) isPublishing(channelID string) bool {
+	c, err := s.GetVideoChannelStatue(channelID)
+	if !err {
+		return false
+	}
+
+	if p, err := s.signal.GetStreamStatus(c.Ssrc); err != nil || !p {
+		return false
+	}
+	return true
+}
+
+func (s *UAS) Bye() error {
+	return nil
+}
+
+func (s *UAS) Catalog(deviceID string) error {
+	var CatalogXML = `<?xml version="1.0"?><Query>
+	<CmdType>Catalog</CmdType>
+	<SN>%d</SN>
+	<DeviceID>%s</DeviceID>
+	</Query>
+	`
+
+	d, ok := DM.GetDevice(deviceID)
+	if !ok {
+		return errors.Errorf("device %s not found", deviceID)
+	}
+
+	body := fmt.Sprintf(CatalogXML, s.getSN(), deviceID)
+
+	req, err := stack.NewCatelogRequest([]byte(body), stack.OutboundConfig{
+		Via:       d.SourceAddr,
+		To:        d.DeviceID,
+		From:      s.conf.Serial,
+		Transport: d.NetworkType,
+	})
+	if err != nil {
+		return errors.Wrapf(err, "build catalog request error")
+	}
+
+	tx, err := s.sipCli.TransactionRequest(s.ctx, req)
+	if err != nil {
+		return errors.Wrapf(err, "transaction request error")
+	}
+
+	res, err := s.waitAnswer(tx)
+	if err != nil {
+		return errors.Wrapf(err, "wait answer error")
+	}
+	logger.Tf(s.ctx, "catalog response: %s", res.String())
+
+	return nil
+
+}
+
+func (s *UAS) waitAnswer(tx sip.ClientTransaction) (*sip.Response, error) {
+	select {
+	case <-s.ctx.Done():
+		return nil, errors.Errorf("context done")
+	case res := <-tx.Responses():
+		if res.StatusCode == 100 || res.StatusCode == 101 || res.StatusCode == 180 || res.StatusCode == 183 {
+			return s.waitAnswer(tx)
+		}
+		return res, nil
+	}
+}

pkg/service/service.go

@@ -0,0 +1,52 @@
+package service
+
+import (
+	"context"
+
+	"github.com/emiago/sipgo"
+	"github.com/ossrs/srs-sip/pkg/config"
+	"github.com/rs/zerolog"
+)
+
+type Service struct {
+	ctx  context.Context
+	conf *config.MainConfig
+	Uac  *UAC
+	Uas  *UAS
+}
+
+func NewService(ctx context.Context, r0 interface{}) (*Service, error) {
+	s := &Service{
+		ctx:  ctx,
+		conf: r0.(*config.MainConfig),
+	}
+	s.Uac = NewUac()
+	s.Uas = NewUas()
+	return s, nil
+}
+
+func (s *Service) Start() error {
+	zerolog.SetGlobalLevel(zerolog.Disabled)
+
+	ua, err := sipgo.NewUA(
+		sipgo.WithUserAgent(UserAgent),
+	)
+	if err != nil {
+		return err
+	}
+
+	if err := s.Uas.Start(ua, s.conf); err != nil {
+		return err
+	}
+
+	// if err := s.Uac.Start(ua, s.conf); err != nil {
+	// 	return err
+	// }
+
+	return nil
+}
+
+func (s *Service) Stop() {
+	s.Uac.Stop()
+	s.Uas.Stop()
+}

pkg/service/stack/request.go

@@ -0,0 +1,68 @@
+package stack
+
+import (
+	"github.com/emiago/sipgo/sip"
+	"github.com/ossrs/go-oryx-lib/errors"
+)
+
+type OutboundConfig struct {
+	Transport string
+	Via       string
+	From      string
+	To        string
+}
+
+func newRequest(method sip.RequestMethod, body []byte, conf OutboundConfig) (*sip.Request, error) {
+	if len(conf.From) != 20 || len(conf.To) != 20 {
+		return nil, errors.Errorf("From or To length is not 20")
+	}
+
+	dest := conf.Via
+	to := sip.Uri{User: conf.To, Host: conf.To[:10]}
+	from := &sip.Uri{User: conf.From, Host: conf.From[:10]}
+
+	fromHeader := &sip.FromHeader{Address: *from, Params: sip.NewParams()}
+	fromHeader.Params.Add("tag", sip.GenerateTagN(16))
+
+	req := sip.NewRequest(method, to)
+	req.AppendHeader(fromHeader)
+	req.AppendHeader(&sip.ToHeader{Address: to})
+	req.AppendHeader(&sip.ContactHeader{Address: *from})
+	req.AppendHeader(sip.NewHeader("Max-Forwards", "70"))
+	req.SetBody(body)
+	req.SetDestination(dest)
+	req.SetTransport(conf.Transport)
+
+	return req, nil
+}
+
+func NewRegisterRequest(conf OutboundConfig) (*sip.Request, error) {
+	req, err := newRequest(sip.REGISTER, nil, conf)
+	if err != nil {
+		return nil, err
+	}
+	req.AppendHeader(sip.NewHeader("Expires", "3600"))
+
+	return req, nil
+}
+
+func NewInviteRequest(body []byte, subject string, conf OutboundConfig) (*sip.Request, error) {
+	req, err := newRequest(sip.INVITE, body, conf)
+	if err != nil {
+		return nil, err
+	}
+	req.AppendHeader(sip.NewHeader("Content-Type", "application/sdp"))
+	req.AppendHeader(sip.NewHeader("Subject", subject))
+
+	return req, nil
+}
+
+func NewCatelogRequest(body []byte, conf OutboundConfig) (*sip.Request, error) {
+	req, err := newRequest(sip.MESSAGE, body, conf)
+	if err != nil {
+		return nil, err
+	}
+	req.AppendHeader(sip.NewHeader("Content-Type", "Application/MANSCDP+xml"))
+
+	return req, nil
+}

pkg/service/stack/response.go

@@ -0,0 +1,25 @@
+package stack
+
+import (
+	"time"
+
+	"github.com/emiago/sipgo/sip"
+)
+
+const TIME_LAYOUT = "2024-01-01T00:00:00"
+const EXPIRES_TIME = 3600
+
+func NewRegisterResponse(req *sip.Request, code sip.StatusCode, reason string) *sip.Response {
+	resp := sip.NewResponseFromRequest(req, code, reason, nil)
+
+	newTo := &sip.ToHeader{Address: resp.To().Address, Params: sip.NewParams()}
+	newTo.Params.Add("tag", sip.GenerateTagN(10))
+
+	resp.ReplaceHeader(newTo)
+	resp.RemoveHeader("Allow")
+	expires := sip.ExpiresHeader(EXPIRES_TIME)
+	resp.AppendHeader(&expires)
+	resp.AppendHeader(sip.NewHeader("Date", time.Now().Format(TIME_LAYOUT)))
+
+	return resp
+}

pkg/service/uac.go

@@ -0,0 +1,122 @@
+package service
+
+import (
+	"context"
+	"fmt"
+
+	"github.com/emiago/sipgo"
+	"github.com/emiago/sipgo/sip"
+	"github.com/ossrs/go-oryx-lib/errors"
+	"github.com/ossrs/go-oryx-lib/logger"
+	"github.com/ossrs/srs-sip/pkg/config"
+	"github.com/ossrs/srs-sip/pkg/service/stack"
+)
+
+const (
+	UserAgent = "SRS-SIP/1.0"
+)
+
+type UAC struct {
+	*Cascade
+
+	SN      uint32
+	LocalIP string
+}
+
+func NewUac() *UAC {
+	ip, err := config.GetLocalIP()
+	if err != nil {
+		logger.E("get local ip failed")
+		return nil
+	}
+
+	c := &UAC{
+		Cascade: &Cascade{},
+		LocalIP: ip,
+	}
+	return c
+}
+
+func (c *UAC) Start(agent *sipgo.UserAgent, r0 interface{}) error {
+	var err error
+
+	c.ctx = context.Background()
+	c.conf = r0.(*config.MainConfig)
+
+	if agent == nil {
+		ua, err := sipgo.NewUA(sipgo.WithUserAgent(UserAgent))
+		if err != nil {
+			return err
+		}
+		agent = ua
+	}
+
+	c.sipCli, err = sipgo.NewClient(agent, sipgo.WithClientHostname(c.LocalIP))
+	if err != nil {
+		return err
+	}
+
+	c.sipSvr, err = sipgo.NewServer(agent)
+	if err != nil {
+		return err
+	}
+
+	c.sipSvr.OnInvite(c.onInvite)
+	c.sipSvr.OnBye(c.onBye)
+	c.sipSvr.OnMessage(c.onMessage)
+
+	go c.doRegister()
+
+	return nil
+}
+
+func (c *UAC) Stop() {
+	// TODO: 断开所有当前连接
+	c.sipCli.Close()
+	c.sipSvr.Close()
+}
+
+func (c *UAC) doRegister() error {
+	r, _ := stack.NewRegisterRequest(stack.OutboundConfig{
+		From:      "34020000001110000001",
+		To:        "34020000002000000001",
+		Transport: "UDP",
+		Via:       fmt.Sprintf("%s:%d", c.LocalIP, c.conf.SipPort),
+	})
+	tx, err := c.sipCli.TransactionRequest(c.ctx, r)
+	if err != nil {
+		return errors.Wrapf(err, "transaction request error")
+	}
+
+	rs, _ := c.getResponse(tx)
+	logger.Tf(c.ctx, "register response: %s", rs.String())
+	return nil
+}
+
+func (c *UAC) OnRequest(req *sip.Request, tx sip.ServerTransaction) {
+	switch req.Method {
+	case "INVITE":
+		c.onInvite(req, tx)
+	}
+}
+
+func (c *UAC) onInvite(req *sip.Request, tx sip.ServerTransaction) {
+	logger.T(c.ctx, "onInvite")
+}
+
+func (c *UAC) onBye(req *sip.Request, tx sip.ServerTransaction) {
+	logger.T(c.ctx, "onBye")
+}
+
+func (c *UAC) onMessage(req *sip.Request, tx sip.ServerTransaction) {
+	logger.Tf(c.ctx, "onMessage %s", req.String())
+}
+
+func (c *UAC) getResponse(tx sip.ClientTransaction) (*sip.Response, error) {
+	select {
+	case <-tx.Done():
+		return nil, fmt.Errorf("transaction died")
+	case res := <-tx.Responses():
+		return res, nil
+	}
+}

pkg/service/uas.go

@@ -0,0 +1,137 @@
+package service
+
+import (
+	"context"
+	"errors"
+	"fmt"
+	"net"
+	"sync"
+
+	"github.com/emiago/sipgo"
+	"github.com/emiago/sipgo/sip"
+	"github.com/ossrs/go-oryx-lib/logger"
+	"github.com/ossrs/srs-sip/pkg/config"
+	"github.com/ossrs/srs-sip/pkg/signaling"
+)
+
+type UAS struct {
+	*Cascade
+
+	SN             uint32
+	channelsStatue sync.Map
+	signal         signaling.ISignaling
+
+	sipConnUDP *net.UDPConn
+	sipConnTCP *net.TCPListener
+}
+
+var DM = GetDeviceManager()
+
+func NewUas() *UAS {
+	return &UAS{
+		Cascade: &Cascade{},
+	}
+}
+
+func (s *UAS) Start(agent *sipgo.UserAgent, r0 interface{}) error {
+	ctx := context.Background()
+	conf := r0.(*config.MainConfig)
+	sig := &signaling.Srs{
+		Ctx:  ctx,
+		Addr: "http://" + conf.MediaAddr,
+	}
+	s.signal = sig
+	s.startSipServer(agent, ctx, r0)
+	return nil
+}
+
+func (s *UAS) Stop() {
+	s.sipCli.Close()
+	s.sipSvr.Close()
+}
+
+func (s *UAS) startSipServer(agent *sipgo.UserAgent, ctx context.Context, r0 interface{}) error {
+	conf := r0.(*config.MainConfig)
+	s.ctx = ctx
+	s.conf = conf
+
+	if agent == nil {
+		ua, err := sipgo.NewUA(sipgo.WithUserAgent(UserAgent))
+		if err != nil {
+			return err
+		}
+		agent = ua
+	}
+
+	cli, err := sipgo.NewClient(agent)
+	if err != nil {
+		return err
+	}
+	s.sipCli = cli
+
+	svr, err := sipgo.NewServer(agent)
+	if err != nil {
+		return err
+	}
+	s.sipSvr = svr
+
+	s.sipSvr.OnRegister(s.onRegister)
+	s.sipSvr.OnMessage(s.onMessage)
+	s.sipSvr.OnNotify(s.onNotify)
+
+	if err := s.startUDP(); err != nil {
+		return err
+	}
+	if err := s.startTCP(); err != nil {
+		return err
+	}
+
+	return nil
+}
+
+func (s *UAS) startUDP() error {
+	lis, err := net.ListenUDP("udp", &net.UDPAddr{
+		IP:   net.IPv4(0, 0, 0, 0),
+		Port: s.conf.SipPort,
+	})
+	if err != nil {
+		return fmt.Errorf("cannot listen on the UDP signaling port %d: %w", s.conf.SipPort, err)
+	}
+	s.sipConnUDP = lis
+	logger.Tf(s.ctx, "sip signaling listening on UDP %s:%d", lis.LocalAddr().String(), s.conf.SipPort)
+
+	go func() {
+		if err := s.sipSvr.ServeUDP(lis); err != nil {
+			panic(fmt.Errorf("SIP listen UDP error: %w", err))
+		}
+	}()
+	return nil
+}
+
+func (s *UAS) startTCP() error {
+	lis, err := net.ListenTCP("tcp", &net.TCPAddr{
+		IP:   net.IPv4(0, 0, 0, 0),
+		Port: s.conf.SipPort,
+	})
+	if err != nil {
+		return fmt.Errorf("cannot listen on the TCP signaling port %d: %w", s.conf.SipPort, err)
+	}
+	s.sipConnTCP = lis
+	logger.Tf(s.ctx, "sip signaling listening on TCP %s:%d", lis.Addr().String(), s.conf.SipPort)
+
+	go func() {
+		if err := s.sipSvr.ServeTCP(lis); err != nil && !errors.Is(err, net.ErrClosed) {
+			panic(fmt.Errorf("SIP listen TCP error: %w", err))
+		}
+	}()
+	return nil
+}
+
+func sipErrorResponse(tx sip.ServerTransaction, req *sip.Request) {
+	_ = tx.Respond(sip.NewResponseFromRequest(req, 400, "", nil))
+}
+
+func (s *UAS) getSN() uint32 {
+	s.SN++
+	return s.SN
+}

pkg/signaling/signaling.go

@@ -0,0 +1,75 @@
+package signaling
+
+import (
+	"bytes"
+	"context"
+	"encoding/json"
+	"io"
+	"net/http"
+	"time"
+
+	"github.com/ossrs/go-oryx-lib/errors"
+	"github.com/ossrs/go-oryx-lib/logger"
+)
+
+type ISignaling interface {
+	Publish(id, ssrc string) (int, error)
+	Unpublish(id string) error
+	GetStreamStatus(id string) (bool, error)
+}
+
+// The r is HTTP API to request, like "http://localhost:1985/gb/v1/publish".
+// The req is the HTTP request body, will be marshal to JSON object. nil is no body
+// The res is the HTTP response body, already unmarshal to JSON object.
+func apiRequest(ctx context.Context, r string, req interface{}, res interface{}) error {
+	var buf bytes.Buffer
+	if req != nil {
+		if err := json.NewEncoder(&buf).Encode(req); err != nil {
+			return errors.Wrapf(err, "Marshal body %v", req)
+		}
+	}
+	logger.Tf(ctx, "Request url api=%v with %v bytes", r, buf.Len())
+
+	method := "POST"
+	if req == nil {
+		method = "GET"
+	}
+	reqObj, err := http.NewRequest(method, r, &buf)
+	if err != nil {
+		return errors.Wrapf(err, "HTTP request %v", buf.String())
+	}
+
+	client := &http.Client{Timeout: 10 * time.Second}
+	resObj, err := client.Do(reqObj.WithContext(ctx))
+	if err != nil {
+		return errors.Wrapf(err, "Do HTTP request %v", buf.String())
+	}
+	defer resObj.Body.Close()
+
+	if resObj.StatusCode != http.StatusOK {
+		return errors.Errorf("Server returned status code=%v", resObj.StatusCode)
+	}
+
+	b2, err := io.ReadAll(resObj.Body)
+	if err != nil {
+		return errors.Wrapf(err, "Read response for %v", buf.String())
+	}
+	logger.Tf(ctx, "Response from %v is %v bytes", r, len(b2))
+
+	errorCode := struct {
+		Code int `json:"code"`
+	}{}
+	if err := json.Unmarshal(b2, &errorCode); err != nil {
+		return errors.Wrapf(err, "Unmarshal %v", string(b2))
+	}
+	if errorCode.Code != 0 {
+		return errors.Errorf("Server fail code=%v %v", errorCode.Code, string(b2))
+	}
+
+	if err := json.Unmarshal(b2, res); err != nil {
+		return errors.Wrapf(err, "Unmarshal %v", string(b2))
+	}
+	logger.Tf(ctx, "Parse response to code=%v ok, %v", errorCode.Code, res)
+
+	return nil
+}

pkg/signaling/srs.go

@@ -0,0 +1,95 @@
+package signaling
+
+import (
+	"context"
+
+	"github.com/ossrs/go-oryx-lib/errors"
+)
+
+type Srs struct {
+	Ctx  context.Context
+	Addr string // The address of SRS, eg: http://localhost:1985
+}
+
+func (s *Srs) Publish(id, ssrc string) (int, error) {
+	req := struct {
+		Id   string `json:"id"`
+		SSRC string `json:"ssrc"`
+	}{
+		id, ssrc,
+	}
+
+	res := struct {
+		Code int `json:"code"`
+		Port int `json:"port"`
+	}{}
+
+	if err := apiRequest(s.Ctx, s.Addr+"/gb/v1/publish/", req, &res); err != nil {
+		return 0, errors.Wrapf(err, "gb/v1/publish")
+	}
+
+	return res.Port, nil
+}
+
+func (s *Srs) Unpublish(id string) error {
+	return nil
+}
+
+//	{
+//	    "code": 0,
+//	    "server": "vid-y19n6nm",
+//	    "service": "382k456r",
+//	    "pid": "9495",
+//	    "streams": [{
+//	        "id": "vid-9y0ozy0",
+//	        "name": "0551954854",
+//	        "vhost": "vid-v2ws53u",
+//	        "app": "live",
+//	        "tcUrl": "webrtc://127.0.0.1:1985/live",
+//	        "url": "/live/0551954854",
+//	        "live_ms": 1720428680003,
+//	        "clients": 1,
+//	        "frames": 8431,
+//	        "send_bytes": 66463941,
+//	        "recv_bytes": 89323998,
+//	        "kbps": {
+//	            "recv_30s": 0,
+//	            "send_30s": 0
+//	        },
+//	        "publish": {
+//	            "active": false,
+//	            "cid": "b3op069g"
+//	        },
+//	        "video": null,
+//	        "audio": null
+//	    }]
+//	}
+func (s *Srs) GetStreamStatus(id string) (bool, error) {
+	type Stream struct {
+		Id      string `json:"id"`
+		Name    string `json:"name"`
+		Publish struct {
+			Active bool   `json:"active"`
+			Cid    string `json:"cid"`
+		} `json:"publish"`
+	}
+	res := struct {
+		Code    int      `json:"code"`
+		Streams []Stream `json:"streams"`
+	}{}
+
+	if err := apiRequest(s.Ctx, s.Addr+"/api/v1/streams?count=99", nil, &res); err != nil {
+		return false, errors.Wrapf(err, "api/v1/stream")
+	}
+
+	if len(res.Streams) == 0 {
+		return false, nil
+	} else {
+		for _, v := range res.Streams {
+			if v.Name == id {
+				return v.Publish.Active, nil
+			}
+		}
+	}
+	return false, nil
+}

pkg/utils/utils.go

@@ -0,0 +1,66 @@
+package utils
+
+import (
+	"context"
+	"crypto/rand"
+	"flag"
+	"math/big"
+	"os"
+
+	"github.com/ossrs/srs-sip/pkg/config"
+)
+
+func Parse(ctx context.Context) interface{} {
+	fl := flag.NewFlagSet(os.Args[0], flag.ContinueOnError)
+
+	var conf config.MainConfig
+	fl.StringVar(&conf.Serial, "serial", "34020000002000000001", "The serial number")
+	fl.StringVar(&conf.Realm, "realm", "3402000000", "The realm")
+	fl.StringVar(&conf.SipHost, "sip-host", "0.0.0.0", "The SIP host")
+	fl.IntVar(&conf.SipPort, "sip-port", 5060, "The SIP port")
+	fl.StringVar(&conf.MediaAddr, "media-addr", "127.0.0.1:1985", "The api address of media server. like: 127.0.0.1:1985")
+	fl.IntVar(&conf.HttpServerPort, "http-server-port", 8888, "The port of http server")
+	fl.IntVar(&conf.APIPort, "api-port", 2020, "The port of http api server")
+
+	fl.Usage = func() {
+		fl.PrintDefaults()
+	}
+
+	if err := fl.Parse(os.Args[1:]); err == flag.ErrHelp {
+		os.Exit(0)
+	}
+
+	showHelp := conf.MediaAddr == ""
+	if showHelp {
+		fl.Usage()
+		os.Exit(-1)
+	}
+
+	return &conf
+}
+
+func GenRandomNumber(n int) string {
+	var result string
+	for i := 0; i < n; i++ {
+		randomDigit, _ := rand.Int(rand.Reader, big.NewInt(10))
+		result += randomDigit.String()
+	}
+	return result
+}
+
+func CreateSSRC(isLive bool) string {
+	ssrc := make([]byte, 10)
+	if isLive {
+		ssrc[0] = '0'
+	} else {
+		ssrc[0] = '1'
+	}
+	copy(ssrc[1:], GenRandomNumber(9))
+	return string(ssrc)
+}
+
+// @see GB/T28181—2016 附录D 统一编码规则
+func IsVideoChannel(channelID string) bool {
+	deviceType := channelID[10:13]
+	return deviceType == "131" || deviceType == "132"
+}

tools/main.go

@@ -0,0 +1,30 @@
+package main
+
+import (
+	"context"
+	"os"
+	"os/signal"
+	"syscall"
+
+	"github.com/ossrs/go-oryx-lib/logger"
+	"github.com/ossrs/srs-bench/gb28181"
+)
+
+func main() {
+	ctx := context.Background()
+
+	var conf interface{}
+	conf = gb28181.Parse(ctx)
+
+	ctx, cancel := context.WithCancel(ctx)
+	go func() {
+		sigs := make(chan os.Signal, 1)
+		signal.Notify(sigs, syscall.SIGTERM, syscall.SIGINT)
+		for sig := range sigs {
+			logger.Wf(ctx, "Quit for signal %v", sig)
+			cancel()
+		}
+	}()
+
+	gb28181.Run(ctx, conf)
+}

web/html/index.html

@@ -0,0 +1,865 @@
+<!DOCTYPE html>
+<html>
+<head>
+    <title>DEMO</title>   
+    <meta charset="utf-8">
+    <link rel="stylesheet" type="text/css" href="css/bootstrap.min.css"/>
+    <style>
+        body{
+            padding-top: 30px;
+        }
+        #my_modal_footer {
+            margin-top: -20px;
+            padding-top: 3px;
+        }
+        #main_modal {
+            margin-top: -60px;
+        }
+        #rtc_player_modal {
+            margin-top: -60px;
+        }
+        .div_play_time {
+            margin-top: 10px;
+        }
+        #pb_buffer_bg {
+            margin-top: -4px;
+            margin-bottom: 10px;
+        }
+    </style>
+</head>
+<body>
+<img src=' '/>
+<div class="navbar navbar-fixed-top">
+    <div class="navbar-inner">
+        <div class="container">
+            <a id="srs_index" class="brand" href="#">DEMO</a>
+            <div class="nav-collapse collapse">
+                <ul class="nav">
+                    <li class="active" ><a id="nav_gb28181" href="srs_gb28181.html">GB28181</a></li>
+                    <li>
+                        <a href="https://github.com/ossrs/srs">
+                            <img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/srs?style=social">
+                        </a>
+                    </li>
+                </ul>
+            </div>
+        </div>
+    </div>
+</div>
+<div class="container">
+    <div name="detect_flash">
+        <div id="main_flash_alert" class="alert alert-danger fade in hide">
+            <button type="button" class="close" data-dismiss="alert">×</button>
+            <strong><p>Usage:</p></strong>
+            <p>
+                请点击下面的图标，启用Flash
+            </p>
+            <p>
+                若没有见到这个图标，Chrome浏览器请打开
+                <span class="text-info">chrome://settings/content/flash</span> 并修改为"Ask first"。
+            </p>
+        </div>
+        <div id="main_flash_hdr" class="hide">
+        </div>
+    </div>
+
+    <div class="form-inline">
+        API地址与端口
+        <input type="text" id="txt_api_url" class="input-xxlarge">
+        <p></p>
+      
+    </div>
+
+    <div>
+        <div class="row">
+            <div class="span4" id="divSipSessionList">
+                <span>设备列表</span><label id="lab_sip_session"></label>
+                <div style="overflow:scroll; height:330px; width:310px">
+                 <ul id="channelList"></ul>
+                </div>
+                
+
+            </div>
+            <div class="span8">
+                <button class="btn btn-primary" id="btn_query_channel">获取设备</button>
+                <button class="btn btn-primary" id="btn_sip_bye">bye</button>
+                <button class="btn btn-primary" id="btn_sip_querycatalog" style="display:none;">querycatalog</button>
+                <div id="context2">
+                  <div>
+                    <pre id="video"  style="overflow:scroll; height:280px"></pre>
+                  </div>
+               </div>
+            </div>
+        </div>
+      </div>
+      <p></p>
+
+      <div>
+        <div class="row">
+            <div class="span4" id="divMediaChannelList">             
+            </div>
+            <div class="span8">
+                <div id="context2">
+                  URL:<a id="gb28181ChannelId"></a>
+                  <div>
+                    <textarea class="span6" id="txt_rtc_url" rows="2"></textarea>
+                    <button class="btn btn-primary" id="btn_rtc_play">RTC播放</button>
+                  </div>
+                  <div>
+                    <div>Message</div>
+                    <pre id="apiMessage"  style="overflow:scroll; height:210px"></pre>
+                  </div>
+               </div>
+            </div>
+        </div>
+      </div>
+
+      <div id="main_content" class="hide">
+        <div id="main_modal" class="modal hide fade">
+            <div class="modal-header">
+                <button type="button" class="close" data-dismiss="modal" aria-hidden="true">×</button>
+                <h3><a href="https://github.com/ossrs/srs">SrsFlvPlayer</a></h3>
+            </div>
+            <div class="modal-body">
+                <div>
+                    <video id="video_player" width="98%" autoplay controls></video>
+                </div>
+            </div>
+            <div class="modal-footer" id="my_modal_footer">
+                <div>
+                    <div class="btn-group dropup">
+                        <button class="btn btn-primary" id="btn_ptz_up"> 上↑ </button>
+                        <button class="btn btn-primary" id="btn_ptz_down"> 下↓ </button>
+                        <button class="btn btn-primary" id="btn_ptz_left"> ←左 </button>
+                        <button class="btn btn-primary" id="btn_ptz_right"> 右→ </button>
+                        <button class="btn btn-primary" id="btn_ptz_zoomin"> 放大＋ </button>
+                        <button class="btn btn-primary" id="btn_ptz_zoomout"> 缩小－ </button>
+                    </div>    
+                </div>
+            </div>
+        </div>
+
+        <div id="rtc_player_modal" class="modal hide fade">
+            <div class="modal-header">
+                <button type="button" class="close" data-dismiss="modal" aria-hidden="true">×</button>
+                <h3><a href="https://github.com/ossrs/srs">RtcPlayer</a></h3>
+            </div>
+            <div class="modal-body">
+                <video id="rtc_media_player" width="100%" controls autoplay ></video>
+            </div>
+            <div class="modal-footer" id="my_modal_footer">
+                <div>
+                    <div class="btn-group dropup">
+                        <button class="btn btn-primary" id="btn_ptz_up_rtc"> 上↑ </button>
+                        <button class="btn btn-primary" id="btn_ptz_down_rtc"> 下↓ </button>
+                        <button class="btn btn-primary" id="btn_ptz_left_rtc"> ←左 </button>
+                        <button class="btn btn-primary" id="btn_ptz_right_rtc"> 右→ </button>
+                        <button class="btn btn-primary" id="btn_ptz_zoomin_rtc"> 放大＋ </button>
+                        <button class="btn btn-primary" id="btn_ptz_zoomout_rtc"> 缩小－ </button>
+                    </div>    
+                </div>
+            </div>
+        </div>
+
+    </div>
+
+    <footer>
+        <p></p>
+        <p><a href="https://github.com/ossrs/srs">SRS Team &copy; 2013</a></p>
+    </footer>
+</div>
+</body>
+<script type="text/javascript" src="js/jquery-1.12.2.min.js"></script>
+<script type="text/javascript" src="js/bootstrap.min.js"></script>
+<script type="text/javascript" src="js/json2.js"></script>
+<script type="text/javascript" src="js/srs.page.js"></script>
+<script type="text/javascript" src="js/srs.log.js"></script>
+<script type="text/javascript" src="js/srs.utility.js"></script>
+<script type="text/javascript" src="js/winlin.utility.js"></script>
+<script type="text/javascript">
+    $(function(){
+        $('#main_content').show();
+        autoLoadPage();
+    });
+</script>
+<script type="text/javascript">
+    var srs_player = null;
+    var url = null;
+
+    var query = parse_query_string();
+    var query_host = query.host.split(':');
+    if (query_host && query_host.length == 2) {
+        $("#txt_api_url").val("http://" + query_host[0] + ":2020");
+    } else {
+        $("#txt_api_url").val("http://" + query.host + ":2020");
+    }
+    var __active_dar = null;
+    function select_dar(dar_id, num, den) {
+        srs_player.set_dar(num, den);
+
+        if (__active_dar) {
+            __active_dar.removeClass("active");
+        }
+
+        __active_dar = $(dar_id).parent();
+        __active_dar.addClass("active");
+    }
+
+    var __active_size = null;
+    function select_fs_size(size_id, refer, percent) {
+        srs_player.set_fs(refer, percent);
+
+        if (__active_size) {
+            __active_size.removeClass("active");
+        }
+
+        __active_size = $(size_id).parent();
+        __active_size.addClass("active");
+    }
+
+    function select_buffer(buffer_time) {
+        var bt = buffer_time;
+        var bt_id = "#btn_bt_" + bt.toFixed(1).replace(".", "_");
+        select_buffer_time(bt_id, bt);
+    }
+    function select_max_buffer(max_buffer_time) {
+        var mbt = max_buffer_time;
+        var mbt_id = "#btn_mbt_" + mbt.toFixed(1).replace(".", "_");
+        select_max_buffer_time(mbt_id, mbt);
+    }
+
+    var __active_bt = null;
+    function select_buffer_time(bt_id, buffer_time) {
+        srs_player.set_bt(buffer_time);
+
+        if (__active_bt) {
+            __active_bt.removeClass("active");
+        }
+
+        __active_bt = $(bt_id).parent();
+        __active_bt.addClass("active");
+
+        select_max_buffer(srs_player.max_buffer_time);
+    }
+
+    var __active_mbt = null;
+    function select_max_buffer_time(mbt_id, max_buffer_time) {
+        srs_player.set_mbt(max_buffer_time);
+
+        if (__active_mbt) {
+            __active_mbt.removeClass("active");
+        }
+
+        __active_mbt = $(mbt_id).parent();
+        __active_mbt.addClass("active");
+    }
+
+
+    //格式化json显示
+    function syntaxHighlight(json) {
+        if (typeof json != 'string') {
+        json = JSON.stringify(json, undefined, 2);
+        }
+        json = json.replace(/&/g, '&').replace(/</g, '<').replace(/>/g, '>');
+        return json.replace(/("(\\u[a-zA-Z0-9]{4}|\\[^u]|[^\\"])*"(\s*:)?|\b(true|false|null)\b|-?\d+(?:\.\d*)?(?:[eE][+\-]?\d+)?)/g, function(match) {
+        var cls = 'number';
+        if (/^"/.test(match)) {
+            if (/:$/.test(match)) {
+                cls = 'key';
+            } else {
+            cls = 'string';
+            }
+        } else if (/true|false/.test(match)) {
+            cls = 'boolean';
+        } else if (/null/.test(match)) {
+            cls = 'null';
+        }
+        return '<span class="' + cls + '">' + match + '</span>';
+        });
+    }  
+
+    function http_get(url){
+        var retdata = null;
+        console.log("GET", url);
+        $.ajax({
+            type       : "GET",
+            async      : false,
+            url        : url,
+            contentType: "text/html",
+            data       : "",
+            complete   : function() {
+            },
+            error      : function(ret) {
+                alert("GET 请求失败:" + url);
+            },
+            success    : function(ret) {
+                console.log(ret);
+                retdata = ret;
+            }
+        });
+        return retdata;
+    }
+
+    function http_post(url, data){
+        var retdata = null;
+        console.log("POST", url);
+        $.ajax({
+            type       : "POST",
+            async      : false,
+            url        : url,
+            contentType: "application/json",
+            data       : JSON.stringify(data),
+            complete   : function() {
+            },
+            error      : function(ret) {
+                alert("POST 请求失败:" + url);
+            },
+            success    : function(ret) {
+                console.log(ret);
+                retdata = ret;
+            }
+        });
+        return retdata;
+    }
+
+    function getParentIdById(id) {
+        for (let parentId in devices) {
+            let parentDevices = devices[parentId];
+            for (let idx in parentDevices) {
+                if (parentDevices[idx].id == id) {
+                    return parentId;
+                }
+            }
+        }
+        return null;
+    }
+
+
+    // request: {"device_id": "1", "channel_id": "1", "sub_stream": 0}
+    // response: {"code": 0, "data": {"channel_id": "1", "url": "webrtc://"}}
+    function channelOnClick(chidObj){
+        var chId = chidObj.text;
+        var parentId = getParentIdById(chId); 
+        
+        var body = {
+            "device_id": parentId,
+            "channel_id": chId,
+            "sub_stream": "0"
+        };
+        url = $("#txt_api_url").val();
+        var apiurl = url + "/srs-sip/v1/invite";
+        
+        var ret = http_post(apiurl, body);
+        $('#sipSessionMessage').html(syntaxHighlight(ret)); 
+
+        if (ret == undefined || ret.code != 0) {
+            alert("invite请求失败");
+            return;
+        }
+        play(ret.data);
+    } 
+
+    function play(data) {
+        $("#txt_rtc_url").val(data.url);
+        $("#btn_rtc_play").click();
+    }
+
+    function refreshGb28181ChList(data) {
+        $("#channelList").empty();
+        devices = {};
+
+        // 遍历数据，将设备信息按照parent_id进行分组
+        for (let idx in data) {
+            var device = data[idx];
+            var parent = device.parent_id;
+            var id = device.device_id;
+
+            // 如果该parent_id不存在在devices对象中，则创建一个新数组
+            if (!devices[parent]) {
+                devices[parent] = [];
+            }
+
+            // 将设备信息添加到对应的parent_id数组中
+            devices[parent].push({ id: id, ip: device.ip_address, name: device.name});
+        }
+
+        // 遍历devices对象，生成设备树
+        for (let parent in devices) {
+            var parentDevices = devices[parent];
+            var parentLi = "<li>" + parent + "<ul>";
+
+            // 遍历该parent_id下的设备信息
+            parentDevices.forEach(function(device) {
+                var id = device.id;
+                var title = device.name + "(" + device.ip + ")";
+                var childLi = "<li><a id='linkChannelId" + id + "' href='javascript:void(0)' title='" + title + "' onclick='channelOnClick(this)'>" + id + "</a></li>";
+                parentLi += childLi;
+            });
+
+            parentLi += "</ul></li>";
+
+            // 将生成的设备树添加到页面中
+            $("#channelList").append(parentLi);
+        }
+
+    }
+
+    // Async-await-promise based SRS RTC Player.
+    function SrsRtcPlayerAsync() {
+        var self = {};
+
+        // @see https://github.com/rtcdn/rtcdn-draft
+        // @url The WebRTC url to play with, for example:
+        //      webrtc://r.ossrs.net/live/livestream
+        // or specifies the API port:
+        //      webrtc://r.ossrs.net:11985/live/livestream
+        // or autostart the play:
+        //      webrtc://r.ossrs.net/live/livestream?autostart=true
+        // or change the app from live to myapp:
+        //      webrtc://r.ossrs.net:11985/myapp/livestream
+        // or change the stream from livestream to mystream:
+        //      webrtc://r.ossrs.net:11985/live/mystream
+        // or set the api server to myapi.domain.com:
+        //      webrtc://myapi.domain.com/live/livestream
+        // or set the candidate(ip) of answer:
+        //      webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
+        // or force to access https API:
+        //      webrtc://r.ossrs.net/live/livestream?schema=https
+        // or use plaintext, without SRTP:
+        //      webrtc://r.ossrs.net/live/livestream?encrypt=false
+        // or any other information, will pass-by in the query:
+        //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
+        //      webrtc://r.ossrs.net/live/livestream?token=xxx
+        self.play = async function(url) {
+            var conf = self.__internal.prepareUrl(url);
+            self.pc.addTransceiver("audio", {direction: "recvonly"});
+            self.pc.addTransceiver("video", {direction: "recvonly"});
+
+            var offer = await self.pc.createOffer();
+            await self.pc.setLocalDescription(offer);
+            var session = await new Promise(function(resolve, reject) {
+                // @see https://github.com/rtcdn/rtcdn-draft
+                var data = {
+                    api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
+                };
+                console.log("Generated offer: ", data);
+
+                $.ajax({
+                    type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
+                    contentType:'application/json', dataType: 'json'
+                }).done(function(data) {
+                    console.log("Got answer: ", data);
+                    if (data.code) {
+                        reject(data); return;
+                    }
+
+                    resolve(data);
+                }).fail(function(reason){
+                    reject(reason);
+                });
+            });
+            await self.pc.setRemoteDescription(
+                new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+            );
+            return session;
+        };
+
+        // Close the publisher.
+        self.close = function() {
+            self.pc.close();
+        };
+
+        // The callback when got remote stream.
+        self.onaddstream = function (event) {};
+
+        // Internal APIs.
+        self.__internal = {
+            defaultPath: '/rtc/v1/play/',
+            prepareUrl: function (webrtcUrl) {
+                var urlObject = self.__internal.parse(webrtcUrl);
+
+                // If user specifies the schema, use it as API schema.
+                var schema = urlObject.user_query.schema;
+                schema = schema ? schema + ':' : window.location.protocol;
+
+                var port = urlObject.port || 1985;
+                if (schema === 'https:') {
+                    port = urlObject.port || 443;
+                }
+
+                // @see https://github.com/rtcdn/rtcdn-draft
+                var api = urlObject.user_query.play || self.__internal.defaultPath;
+                if (api.lastIndexOf('/') !== api.length - 1) {
+                    api += '/';
+                }
+
+                apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+                for (var key in urlObject.user_query) {
+                    if (key !== 'api' && key !== 'play') {
+                        apiUrl += '&' + key + '=' + urlObject.user_query[key];
+                    }
+                }
+                // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
+                var apiUrl = apiUrl.replace(api + '&', api + '?');
+
+                var streamUrl = urlObject.url;
+
+                return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
+            },
+            parse: function (url) {
+                // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
+                var a = document.createElement("a");
+                a.href = url.replace("rtmp://", "http://")
+                    .replace("webrtc://", "http://")
+                    .replace("rtc://", "http://");
+
+                var vhost = a.hostname;
+                var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
+                var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
+
+                // parse the vhost in the params of app, that srs supports.
+                app = app.replace("...vhost...", "?vhost=");
+                if (app.indexOf("?") >= 0) {
+                    var params = app.slice(app.indexOf("?"));
+                    app = app.slice(0, app.indexOf("?"));
+
+                    if (params.indexOf("vhost=") > 0) {
+                        vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+                        if (vhost.indexOf("&") > 0) {
+                            vhost = vhost.slice(0, vhost.indexOf("&"));
+                        }
+                    }
+                }
+
+                // when vhost equals to server, and server is ip,
+                // the vhost is __defaultVhost__
+                if (a.hostname === vhost) {
+                    var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
+                    if (re.test(a.hostname)) {
+                        vhost = "__defaultVhost__";
+                    }
+                }
+
+                // parse the schema
+                var schema = "rtmp";
+                if (url.indexOf("://") > 0) {
+                    schema = url.slice(0, url.indexOf("://"));
+                }
+
+                var port = a.port;
+                if (!port) {
+                    if (schema === 'http') {
+                        port = 80;
+                    } else if (schema === 'https') {
+                        port = 443;
+                    } else if (schema === 'rtmp') {
+                        port = 1935;
+                    }
+                }
+
+                var ret = {
+                    url: url,
+                    schema: schema,
+                    server: a.hostname, port: port,
+                    vhost: vhost, app: app, stream: stream
+                };
+                self.__internal.fill_query(a.search, ret);
+
+                // For webrtc API, we use 443 if page is https, or schema specified it.
+                if (!ret.port) {
+                    if (schema === 'webrtc' || schema === 'rtc') {
+                        if (ret.user_query.schema === 'https') {
+                            ret.port = 443;
+                        } else if (window.location.href.indexOf('https://') === 0) {
+                            ret.port = 443;
+                        } else {
+                            // For WebRTC, SRS use 1985 as default API port.
+                            ret.port = 1985;
+                        }
+                    }
+                }
+
+                return ret;
+            },
+            fill_query: function (query_string, obj) {
+                // pure user query object.
+                obj.user_query = {};
+
+                if (query_string.length === 0) {
+                    return;
+                }
+
+                // split again for angularjs.
+                if (query_string.indexOf("?") >= 0) {
+                    query_string = query_string.split("?")[1];
+                }
+
+                var queries = query_string.split("&");
+                for (var i = 0; i < queries.length; i++) {
+                    var elem = queries[i];
+
+                    var query = elem.split("=");
+                    obj[query[0]] = query[1];
+                    obj.user_query[query[0]] = query[1];
+                }
+
+                // alias domain for vhost.
+                if (obj.domain) {
+                    obj.vhost = obj.domain;
+                }
+            }
+        };
+
+        self.pc = new RTCPeerConnection(null);
+        self.pc.onaddstream = function (event) {
+            if (self.onaddstream) {
+                self.onaddstream(event);
+            }
+        };
+
+        return self;
+    }
+
+    var flvPlayer = null;
+    var hlsPlayer = null;
+    var devices = {};
+
+    var stopPlayers = function () {
+        if (flvPlayer) {
+            flvPlayer.destroy();
+            flvPlayer = null;
+        }
+        if (hlsPlayer) {
+            hlsPlayer.destroy();
+            hlsPlayer = null;
+        }
+    };
+
+    var hide_for_error = function () {
+        $('#main_flash_alert').show();
+        $('#main_info').hide();
+        $('#main_tips').hide();
+        $('#video_player').hide();
+        //$('#btn_play').hide();
+
+        stopPlayers();
+    };
+
+    var show_for_ok = function () {
+        $('#main_flash_alert').hide();
+        $('#main_info').show();
+        $('#main_tips').show();
+        $('#video_player').show();
+        //$('#btn_play').show();
+    };
+
+
+    /****
+    * The parameters for this page:
+    *       schema, the protocol schema, rtmp or http.
+    *       server, the ip of the url.
+    *       port, the rtmp port of url.
+    *       vhost, the vhost of url, can equals to server.
+    *       app, the app of url.
+    *       stream, the stream of url, can endwith .flv or .mp4 or nothing for RTMP.
+    *       autostart, whether auto play the stream.
+    *       buffer, the buffer time in seconds.
+    * extra params:
+    *       shp_identify, hls+ param.
+    * for example:
+    *       http://localhost:8088/players/srs_player.html?vhost=ossrs.net&app=live&stream=livestream&server=ossrs.net&port=1935&autostart=true&schema=rtmp
+    *       http://localhost:8088/players/srs_player.html?vhost=ossrs.net&app=live&stream=livestream.flv&server=ossrs.net&port=8080&autostart=true&schema=http
+    */
+    var autoLoadPage = function() {
+        var query = parse_query_string();
+
+        // get the vhost and port to set the default url.
+        // url set to: http://localhost:8080/live/livestream.flv
+        srs_init_flv("#txt_url", "#main_modal");
+        srs_init_flv("#txt_url", "#rtc_player_modal");
+
+        // consts for buffer and max buffer.
+        var bts =  [0.1, 0.2, 0.3, 0.5, 0.8, 1, 2, 3, 4, 5, 6, 8, 10, 15, 20, 30];
+        var mbts = [0.6, 0.9, 1.2, 1.5, 2.4, 3, 6, 9, 12, 15, 18, 24, 30, 45, 60, 90];
+
+        // the play startup time.
+        var pst = new Date();
+        
+        $("#main_modal").on("show", function(){
+        });
+
+        $("#main_modal").on("hide", function(){
+           
+        });
+
+        if (true) {
+            for (var bt of bts) {
+                var bt_id = "#btn_bt_" + bt.toFixed(1).replace(".", "_");
+
+                var bt_fun = function(id, v){
+                    $(bt_id).click(function(){
+                        select_buffer_time(id, v);
+
+                        // remember the chagned buffer.
+                        if (Number(query.buffer) != srs_player.buffer_time) {
+                            query.buffer = srs_player.buffer_time;
+                            apply_url_change();
+                        }
+
+                    });
+                };
+                bt_fun(bt_id, bt);
+            }
+        }
+        if (true) {
+            for (var mbt of mbts) {
+                var mbt_id = "#btn_mbt_" + mbt.toFixed(1).replace(".", "_");
+
+                var mbt_fun = function(id, v){
+                    $(mbt_id).click(function(){
+                        select_max_buffer_time(id, v);
+                    });
+                };
+                mbt_fun(mbt_id, mbt);
+            }
+        }
+
+        if (true){
+            var time_query = function(){
+                $("#btn_query_channel").click();
+                setTimeout(function () {$("#btn_query_channel").click()}, 1000);
+            };
+
+            $("#btn_sip_bye").click(function(){
+                var text = $("#sipSessionId").text();
+                if (text.indexOf("-->") != -1) {
+                    var str = text.split("-->");
+                    id = str[0];
+                    var str2 = str[1].split(":")
+                    chid = str2[1];
+
+                    url = $("#txt_api_url").val();
+                    var apiurl = url + "/srs-sip/v1/gb28181?action=sip_bye&id=" + id + "&chid="+chid;
+                    var ret = http_get(apiurl);
+                    $('#sipSessionMessage').html(syntaxHighlight(ret)); 
+
+                    if (ret != undefined && ret.code == 0){
+                        time_query();
+                    }
+                }
+            });
+
+            $("#btn_sip_querycatalog").click(function(){
+                var text = $("#sipSessionId").text();
+                if (text.indexOf("-->") != -1) {
+                    var str = text.split("-->");
+                    id = str[0];
+                    var str2 = str[1].split(":")
+                    chid = str2[0];
+
+                    url = $("#txt_api_url").val();
+                    var apiurl = url + "/srs-sip/v1/gb28181?action=sip_query_catalog&id=" + id;
+                    var ret = http_get(apiurl);
+                    $('#sipSessionMessage').html(syntaxHighlight(ret)); 
+                    if (ret != undefined && ret.code == 0){
+                        time_query();
+                    }
+                }
+            });
+
+            $("#btn_query_channel").click(function(){
+                url = $("#txt_api_url").val();
+                var apiurl = url + "/srs-sip/v1/channels/"
+                var ret = http_get(apiurl);
+                $('#apiMessage').html(syntaxHighlight(ret)); 
+
+                if (ret != undefined && ret.code == 0){
+                    refreshGb28181ChList(ret.data);
+                }
+            });
+
+            var call_ptz_cmd = function(cmd) {
+                var str = $("#gb28181ChannelId").text();
+                var str_array = str.split("@")
+                var chid = "";
+                var id = "";
+                if (str_array.length < 1){
+                    return;
+                }
+
+                var speed = "136";
+               
+                id = str_array[0];
+                chid = str_array[1];
+                
+                url = $("#txt_api_url").val();
+                var apiurl = url + "/srs-sip/v1/gb28181?action=sip_ptz&id=" + id + "&chid="+chid+ "&ptzcmd="+cmd + "&speed=" + speed;
+                var ret = http_get(apiurl);
+                $('#apiMessage').html(syntaxHighlight(ret)); 
+            };
+
+            var ptz_cmd = ["up", "down", "right", "left", "zoomin", "zoomout"]
+            for (var i=0; i<ptz_cmd.length; i++){
+                var bt_fun = function(id, cmd){
+                    $(bt_id).mousedown(function(){
+                        call_ptz_cmd(cmd);
+                    });
+                    $(bt_id).mouseup(function(){
+                        call_ptz_cmd("stop");
+                    });
+                };
+
+                var bt_id = "#btn_ptz_"+ptz_cmd[i]+ "_rtc";
+                bt_fun(bt_id, ptz_cmd[i]);
+
+                bt_id = "#btn_ptz_"+ptz_cmd[i];
+                bt_fun(bt_id, ptz_cmd[i]);
+            }
+
+            var sdk = null; // Global handler to do cleanup when replaying.
+            var startPlay = function() {
+                $('#rtc_media_player').show();
+
+                // Close PC when user replay.
+                if (sdk) {
+                    sdk.close();
+                }
+
+                sdk = new SrsRtcPlayerAsync();
+                sdk.onaddstream = function (event) {
+                    console.log('Start play, event: ', event);
+                    $('#rtc_media_player').prop('srcObject', event.stream);
+                };
+
+                // For example:
+                //      webrtc://r.ossrs.net/live/livestream
+                var url = $("#txt_rtc_url").val();
+                sdk.play(url).then(function(session){
+                    $('#sessionid').html(session.sessionid);
+                    $('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
+                }).catch(function (reason) {
+                    sdk.close();
+                    $('#rtc_media_player').hide();
+                    console.error(reason);
+                });
+            };
+
+            $("#btn_rtc_play").click(function(){
+                $('#rtc_media_player').width(srs_get_player_width);
+                $('#rtc_media_player').height(srs_get_player_height);
+                $("#rtc_player_modal").modal({show: true, keyboard: false});
+                
+                startPlay();
+            });
+
+
+            $("#rtc_player_modal").on("hide", function(){
+                if (sdk) {
+                    sdk.close();
+                }
+             });
+
+        }
+    };
+</script>
+</html>

web/html/js/json2.js

@@ -0,0 +1,486 @@
+/*
+    json2.js
+    2013-05-26
+
+    Public Domain.
+
+    NO WARRANTY EXPRESSED OR IMPLIED. USE AT YOUR OWN RISK.
+
+    See http://www.JSON.org/js.html
+
+
+    This code should be minified before deployment.
+    See http://javascript.crockford.com/jsmin.html
+
+    USE YOUR OWN COPY. IT IS EXTREMELY UNWISE TO LOAD CODE FROM SERVERS YOU DO
+    NOT CONTROL.
+
+
+    This file creates a global JSON object containing two methods: stringify
+    and parse.
+
+        JSON.stringify(value, replacer, space)
+            value       any JavaScript value, usually an object or array.
+
+            replacer    an optional parameter that determines how object
+                        values are stringified for objects. It can be a
+                        function or an array of strings.
+
+            space       an optional parameter that specifies the indentation
+                        of nested structures. If it is omitted, the text will
+                        be packed without extra whitespace. If it is a number,
+                        it will specify the number of spaces to indent at each
+                        level. If it is a string (such as '\t' or ' '),
+                        it contains the characters used to indent at each level.
+
+            This method produces a JSON text from a JavaScript value.
+
+            When an object value is found, if the object contains a toJSON
+            method, its toJSON method will be called and the result will be
+            stringified. A toJSON method does not serialize: it returns the
+            value represented by the name/value pair that should be serialized,
+            or undefined if nothing should be serialized. The toJSON method
+            will be passed the key associated with the value, and this will be
+            bound to the value
+
+            For example, this would serialize Dates as ISO strings.
+
+                Date.prototype.toJSON = function (key) {
+                    function f(n) {
+                        // Format integers to have at least two digits.
+                        return n < 10 ? '0' + n : n;
+                    }
+
+                    return this.getUTCFullYear()   + '-' +
+                         f(this.getUTCMonth() + 1) + '-' +
+                         f(this.getUTCDate())      + 'T' +
+                         f(this.getUTCHours())     + ':' +
+                         f(this.getUTCMinutes())   + ':' +
+                         f(this.getUTCSeconds())   + 'Z';
+                };
+
+            You can provide an optional replacer method. It will be passed the
+            key and value of each member, with this bound to the containing
+            object. The value that is returned from your method will be
+            serialized. If your method returns undefined, then the member will
+            be excluded from the serialization.
+
+            If the replacer parameter is an array of strings, then it will be
+            used to select the members to be serialized. It filters the results
+            such that only members with keys listed in the replacer array are
+            stringified.
+
+            Values that do not have JSON representations, such as undefined or
+            functions, will not be serialized. Such values in objects will be
+            dropped; in arrays they will be replaced with null. You can use
+            a replacer function to replace those with JSON values.
+            JSON.stringify(undefined) returns undefined.
+
+            The optional space parameter produces a stringification of the
+            value that is filled with line breaks and indentation to make it
+            easier to read.
+
+            If the space parameter is a non-empty string, then that string will
+            be used for indentation. If the space parameter is a number, then
+            the indentation will be that many spaces.
+
+            Example:
+
+            text = JSON.stringify(['e', {pluribus: 'unum'}]);
+            // text is '["e",{"pluribus":"unum"}]'
+
+
+            text = JSON.stringify(['e', {pluribus: 'unum'}], null, '\t');
+            // text is '[\n\t"e",\n\t{\n\t\t"pluribus": "unum"\n\t}\n]'
+
+            text = JSON.stringify([new Date()], function (key, value) {
+                return this[key] instanceof Date ?
+                    'Date(' + this[key] + ')' : value;
+            });
+            // text is '["Date(---current time---)"]'
+
+
+        JSON.parse(text, reviver)
+            This method parses a JSON text to produce an object or array.
+            It can throw a SyntaxError exception.
+
+            The optional reviver parameter is a function that can filter and
+            transform the results. It receives each of the keys and values,
+            and its return value is used instead of the original value.
+            If it returns what it received, then the structure is not modified.
+            If it returns undefined then the member is deleted.
+
+            Example:
+
+            // Parse the text. Values that look like ISO date strings will
+            // be converted to Date objects.
+
+            myData = JSON.parse(text, function (key, value) {
+                var a;
+                if (typeof value === 'string') {
+                    a =
+/^(\d{4})-(\d{2})-(\d{2})T(\d{2}):(\d{2}):(\d{2}(?:\.\d*)?)Z$/.exec(value);
+                    if (a) {
+                        return new Date(Date.UTC(+a[1], +a[2] - 1, +a[3], +a[4],
+                            +a[5], +a[6]));
+                    }
+                }
+                return value;
+            });
+
+            myData = JSON.parse('["Date(09/09/2001)"]', function (key, value) {
+                var d;
+                if (typeof value === 'string' &&
+                        value.slice(0, 5) === 'Date(' &&
+                        value.slice(-1) === ')') {
+                    d = new Date(value.slice(5, -1));
+                    if (d) {
+                        return d;
+                    }
+                }
+                return value;
+            });
+
+
+    This is a reference implementation. You are free to copy, modify, or
+    redistribute.
+*/
+
+/*jslint evil: true, regexp: true */
+
+/*members "", "\b", "\t", "\n", "\f", "\r", "\"", JSON, "\\", apply,
+    call, charCodeAt, getUTCDate, getUTCFullYear, getUTCHours,
+    getUTCMinutes, getUTCMonth, getUTCSeconds, hasOwnProperty, join,
+    lastIndex, length, parse, prototype, push, replace, slice, stringify,
+    test, toJSON, toString, valueOf
+*/
+
+
+// Create a JSON object only if one does not already exist. We create the
+// methods in a closure to avoid creating global variables.
+
+if (typeof JSON !== 'object') {
+    JSON = {};
+}
+
+(function () {
+    'use strict';
+
+    function f(n) {
+        // Format integers to have at least two digits.
+        return n < 10 ? '0' + n : n;
+    }
+
+    if (typeof Date.prototype.toJSON !== 'function') {
+
+        Date.prototype.toJSON = function () {
+
+            return isFinite(this.valueOf())
+                ? this.getUTCFullYear()     + '-' +
+                    f(this.getUTCMonth() + 1) + '-' +
+                    f(this.getUTCDate())      + 'T' +
+                    f(this.getUTCHours())     + ':' +
+                    f(this.getUTCMinutes())   + ':' +
+                    f(this.getUTCSeconds())   + 'Z'
+                : null;
+        };
+
+        String.prototype.toJSON      =
+            Number.prototype.toJSON  =
+            Boolean.prototype.toJSON = function () {
+                return this.valueOf();
+            };
+    }
+
+    var cx = /[\u0000\u00ad\u0600-\u0604\u070f\u17b4\u17b5\u200c-\u200f\u2028-\u202f\u2060-\u206f\ufeff\ufff0-\uffff]/g,
+        escapable = /[\\\"\x00-\x1f\x7f-\x9f\u00ad\u0600-\u0604\u070f\u17b4\u17b5\u200c-\u200f\u2028-\u202f\u2060-\u206f\ufeff\ufff0-\uffff]/g,
+        gap,
+        indent,
+        meta = {    // table of character substitutions
+            '\b': '\\b',
+            '\t': '\\t',
+            '\n': '\\n',
+            '\f': '\\f',
+            '\r': '\\r',
+            '"' : '\\"',
+            '\\': '\\\\'
+        },
+        rep;
+
+
+    function quote(string) {
+
+// If the string contains no control characters, no quote characters, and no
+// backslash characters, then we can safely slap some quotes around it.
+// Otherwise we must also replace the offending characters with safe escape
+// sequences.
+
+        escapable.lastIndex = 0;
+        return escapable.test(string) ? '"' + string.replace(escapable, function (a) {
+            var c = meta[a];
+            return typeof c === 'string'
+                ? c
+                : '\\u' + ('0000' + a.charCodeAt(0).toString(16)).slice(-4);
+        }) + '"' : '"' + string + '"';
+    }
+
+
+    function str(key, holder) {
+
+// Produce a string from holder[key].
+
+        var i,          // The loop counter.
+            k,          // The member key.
+            v,          // The member value.
+            length,
+            mind = gap,
+            partial,
+            value = holder[key];
+
+// If the value has a toJSON method, call it to obtain a replacement value.
+
+        if (value && typeof value === 'object' &&
+                typeof value.toJSON === 'function') {
+            value = value.toJSON(key);
+        }
+
+// If we were called with a replacer function, then call the replacer to
+// obtain a replacement value.
+
+        if (typeof rep === 'function') {
+            value = rep.call(holder, key, value);
+        }
+
+// What happens next depends on the value's type.
+
+        switch (typeof value) {
+        case 'string':
+            return quote(value);
+
+        case 'number':
+
+// JSON numbers must be finite. Encode non-finite numbers as null.
+
+            return isFinite(value) ? String(value) : 'null';
+
+        case 'boolean':
+        case 'null':
+
+// If the value is a boolean or null, convert it to a string. Note:
+// typeof null does not produce 'null'. The case is included here in
+// the remote chance that this gets fixed someday.
+
+            return String(value);
+
+// If the type is 'object', we might be dealing with an object or an array or
+// null.
+
+        case 'object':
+
+// Due to a specification blunder in ECMAScript, typeof null is 'object',
+// so watch out for that case.
+
+            if (!value) {
+                return 'null';
+            }
+
+// Make an array to hold the partial results of stringifying this object value.
+
+            gap += indent;
+            partial = [];
+
+// Is the value an array?
+
+            if (Object.prototype.toString.apply(value) === '[object Array]') {
+
+// The value is an array. Stringify every element. Use null as a placeholder
+// for non-JSON values.
+
+                length = value.length;
+                for (i = 0; i < length; i += 1) {
+                    partial[i] = str(i, value) || 'null';
+                }
+
+// Join all of the elements together, separated with commas, and wrap them in
+// brackets.
+
+                v = partial.length === 0
+                    ? '[]'
+                    : gap
+                    ? '[\n' + gap + partial.join(',\n' + gap) + '\n' + mind + ']'
+                    : '[' + partial.join(',') + ']';
+                gap = mind;
+                return v;
+            }
+
+// If the replacer is an array, use it to select the members to be stringified.
+
+            if (rep && typeof rep === 'object') {
+                length = rep.length;
+                for (i = 0; i < length; i += 1) {
+                    if (typeof rep[i] === 'string') {
+                        k = rep[i];
+                        v = str(k, value);
+                        if (v) {
+                            partial.push(quote(k) + (gap ? ': ' : ':') + v);
+                        }
+                    }
+                }
+            } else {
+
+// Otherwise, iterate through all of the keys in the object.
+
+                for (k in value) {
+                    if (Object.prototype.hasOwnProperty.call(value, k)) {
+                        v = str(k, value);
+                        if (v) {
+                            partial.push(quote(k) + (gap ? ': ' : ':') + v);
+                        }
+                    }
+                }
+            }
+
+// Join all of the member texts together, separated with commas,
+// and wrap them in braces.
+
+            v = partial.length === 0
+                ? '{}'
+                : gap
+                ? '{\n' + gap + partial.join(',\n' + gap) + '\n' + mind + '}'
+                : '{' + partial.join(',') + '}';
+            gap = mind;
+            return v;
+        }
+    }
+
+// If the JSON object does not yet have a stringify method, give it one.
+
+    if (typeof JSON.stringify !== 'function') {
+        JSON.stringify = function (value, replacer, space) {
+
+// The stringify method takes a value and an optional replacer, and an optional
+// space parameter, and returns a JSON text. The replacer can be a function
+// that can replace values, or an array of strings that will select the keys.
+// A default replacer method can be provided. Use of the space parameter can
+// produce text that is more easily readable.
+
+            var i;
+            gap = '';
+            indent = '';
+
+// If the space parameter is a number, make an indent string containing that
+// many spaces.
+
+            if (typeof space === 'number') {
+                for (i = 0; i < space; i += 1) {
+                    indent += ' ';
+                }
+
+// If the space parameter is a string, it will be used as the indent string.
+
+            } else if (typeof space === 'string') {
+                indent = space;
+            }
+
+// If there is a replacer, it must be a function or an array.
+// Otherwise, throw an error.
+
+            rep = replacer;
+            if (replacer && typeof replacer !== 'function' &&
+                    (typeof replacer !== 'object' ||
+                    typeof replacer.length !== 'number')) {
+                throw new Error('JSON.stringify');
+            }
+
+// Make a fake root object containing our value under the key of ''.
+// Return the result of stringifying the value.
+
+            return str('', {'': value});
+        };
+    }
+
+
+// If the JSON object does not yet have a parse method, give it one.
+
+    if (typeof JSON.parse !== 'function') {
+        JSON.parse = function (text, reviver) {
+
+// The parse method takes a text and an optional reviver function, and returns
+// a JavaScript value if the text is a valid JSON text.
+
+            var j;
+
+            function walk(holder, key) {
+
+// The walk method is used to recursively walk the resulting structure so
+// that modifications can be made.
+
+                var k, v, value = holder[key];
+                if (value && typeof value === 'object') {
+                    for (k in value) {
+                        if (Object.prototype.hasOwnProperty.call(value, k)) {
+                            v = walk(value, k);
+                            if (v !== undefined) {
+                                value[k] = v;
+                            } else {
+                                delete value[k];
+                            }
+                        }
+                    }
+                }
+                return reviver.call(holder, key, value);
+            }
+
+
+// Parsing happens in four stages. In the first stage, we replace certain
+// Unicode characters with escape sequences. JavaScript handles many characters
+// incorrectly, either silently deleting them, or treating them as line endings.
+
+            text = String(text);
+            cx.lastIndex = 0;
+            if (cx.test(text)) {
+                text = text.replace(cx, function (a) {
+                    return '\\u' +
+                        ('0000' + a.charCodeAt(0).toString(16)).slice(-4);
+                });
+            }
+
+// In the second stage, we run the text against regular expressions that look
+// for non-JSON patterns. We are especially concerned with '()' and 'new'
+// because they can cause invocation, and '=' because it can cause mutation.
+// But just to be safe, we want to reject all unexpected forms.
+
+// We split the second stage into 4 regexp operations in order to work around
+// crippling inefficiencies in IE's and Safari's regexp engines. First we
+// replace the JSON backslash pairs with '@' (a non-JSON character). Second, we
+// replace all simple value tokens with ']' characters. Third, we delete all
+// open brackets that follow a colon or comma or that begin the text. Finally,
+// we look to see that the remaining characters are only whitespace or ']' or
+// ',' or ':' or '{' or '}'. If that is so, then the text is safe for eval.
+
+            if (/^[\],:{}\s]*$/
+                    .test(text.replace(/\\(?:["\\\/bfnrt]|u[0-9a-fA-F]{4})/g, '@')
+                        .replace(/"[^"\\\n\r]*"|true|false|null|-?\d+(?:\.\d*)?(?:[eE][+\-]?\d+)?/g, ']')
+                        .replace(/(?:^|:|,)(?:\s*\[)+/g, ''))) {
+
+// In the third stage we use the eval function to compile the text into a
+// JavaScript structure. The '{' operator is subject to a syntactic ambiguity
+// in JavaScript: it can begin a block or an object literal. We wrap the text
+// in parens to eliminate the ambiguity.
+
+                j = eval('(' + text + ')');
+
+// In the optional fourth stage, we recursively walk the new structure, passing
+// each name/value pair to a reviver function for possible transformation.
+
+                return typeof reviver === 'function'
+                    ? walk({'': j}, '')
+                    : j;
+            }
+
+// If the text is not JSON parseable, then a SyntaxError is thrown.
+
+            throw new SyntaxError('JSON.parse');
+        };
+    }
+}());

web/html/js/srs.log.js

@@ -0,0 +1,37 @@
+/**
+* log specified, there must be a log element as:
+    <!-- for the log -->
+    <div class="alert alert-info fade in" id="txt_log">
+        <button type="button" class="close" data-dismiss="alert">×</button>
+        <strong><span id="txt_log_title">标题:</span></strong>
+        <span id="txt_log_msg">日志内容</span>
+    </div>
+*/
+var srs_log_disabled = false;
+function info(desc) {
+    if (srs_log_disabled) {
+        return;
+    }
+    
+    $("#txt_log").addClass("alert-info").removeClass("alert-error").removeClass("alert-warn");
+    $("#txt_log_title").text("Info:");
+    $("#txt_log_msg").html(desc);
+}
+function warn(code, desc) {
+    if (srs_log_disabled) {
+        return;
+    }
+    
+    $("#txt_log").removeClass("alert-info").removeClass("alert-error").addClass("alert-warn");
+    $("#txt_log_title").text("Warn:");
+    $("#txt_log_msg").html("code: " + code + ", " + desc);
+}
+function error(code, desc) {
+    if (srs_log_disabled) {
+        return;
+    }
+    
+    $("#txt_log").removeClass("alert-info").addClass("alert-error").removeClass("alert-warn");
+    $("#txt_log_title").text("Error:");
+    $("#txt_log_msg").html("code: " + code + ", " + desc);
+}

web/html/js/srs.page.js

@@ -0,0 +1,182 @@
+
+// to query the swf anti cache.
+function srs_get_version_code() { return "1.33"; }
+
+/**
+* player specified size.
+*/
+function srs_get_player_modal() { return 740; }
+function srs_get_player_width() { return srs_get_player_modal() - 30; }
+function srs_get_player_height() { return srs_get_player_width() * 9 / 19; }
+
+/**
+* update the navigator, add same query string.
+*/
+function update_nav() {
+    $("#nav_srs_player").attr("href", "srs_player.html" + window.location.search);
+    $("#nav_rtc_player").attr("href", "rtc_player.html" + window.location.search);
+    $("#nav_rtc_publisher").attr("href", "rtc_publisher.html" + window.location.search);
+    $("#nav_whip").attr("href", "whip.html" + window.location.search);
+    $("#nav_whep").attr("href", "whep.html" + window.location.search);
+    $("#nav_srs_publisher").attr("href", "srs_publisher.html" + window.location.search);
+    $("#nav_srs_chat").attr("href", "srs_chat.html" + window.location.search);
+    $("#nav_srs_bwt").attr("href", "srs_bwt.html" + window.location.search);
+    $("#nav_vlc").attr("href", "vlc.html" + window.location.search);
+}
+
+// Special extra params, such as auth_key.
+function user_extra_params(query, params, rtc) {
+    var queries = params || [];
+
+    for (var key in query.user_query) {
+        if (key === 'app' || key === 'autostart' || key === 'dir'
+            || key === 'filename' || key === 'host' || key === 'hostname'
+            || key === 'http_port' || key === 'pathname' || key === 'port'
+            || key === 'server' || key === 'stream' || key === 'buffer'
+            || key === 'schema' || key === 'vhost' || key === 'api'
+            || key === 'path'
+        ) {
+            continue;
+        }
+
+        if (query[key]) {
+            queries.push(key + '=' + query[key]);
+        }
+    }
+
+    return queries;
+}
+
+function is_default_port(schema, port) {
+    return (schema === 'http' && port === 80)
+        || (schema === 'https' && port === 443)
+        || (schema === 'webrtc' && port === 1985)
+        || (schema === 'rtmp' && port === 1935);
+}
+
+/**
+@param server the ip of server. default to window.location.hostname
+@param vhost the vhost of HTTP-FLV. default to window.location.hostname
+@param port the port of HTTP-FLV. default to 1935
+@param app the app of HTTP-FLV. default to live.
+@param stream the stream of HTTP-FLV. default to livestream.flv
+*/
+function build_default_flv_url() {
+    var query = parse_query_string();
+
+    var schema = (!query.schema)? "http":query.schema;
+    var server = (!query.server)? window.location.hostname:query.server;
+    var port = (!query.port)? (schema==="http"? 8080:1935) : Number(query.port);
+    var vhost = (!query.vhost)? window.location.hostname:query.vhost;
+    var app = (!query.app)? "live":query.app;
+    var stream = (!query.stream)? "livestream.flv":query.stream;
+
+    var queries = [];
+    if (server !== vhost && vhost !== "__defaultVhost__") {
+        queries.push("vhost=" + vhost);
+    }
+    queries = user_extra_params(query, queries);
+
+    var uri = schema + "://" + server;
+    if (!is_default_port(schema, port)) {
+        uri += ":" + port;
+    }
+    uri += "/" + app + "/" + stream + "?" + queries.join('&');
+    while (uri.indexOf("?") === uri.length - 1) {
+        uri = uri.slice(0, uri.length - 1);
+    }
+
+    return uri;
+}
+
+function build_default_rtc_url(query) {
+    // The format for query string to overwrite configs of server.
+    console.log('?eip=x.x.x.x to overwrite candidate. 覆盖服务器candidate(外网IP)配置');
+    console.log('?api=x to overwrite WebRTC API(1985).');
+    console.log('?schema=http|https to overwrite WebRTC API protocol.');
+
+    var server = (!query.server)? window.location.hostname:query.server;
+    var vhost = (!query.vhost)? window.location.hostname:query.vhost;
+    var app = (!query.app)? "live":query.app;
+    var stream = (!query.stream)? "livestream":query.stream;
+    var api = query.api? ':'+query.api : '';
+
+    var queries = [];
+    if (server !== vhost && vhost !== "__defaultVhost__") {
+        queries.push("vhost=" + vhost);
+    }
+    if (query.schema && window.location.protocol !== query.schema + ':') {
+        queries.push('schema=' + query.schema);
+    }
+    queries = user_extra_params(query, queries, true);
+
+    var uri = "webrtc://" + server + api + "/" + app + "/" + stream + "?" + queries.join('&');
+    while (uri.lastIndexOf("?") === uri.length - 1) {
+        uri = uri.slice(0, uri.length - 1);
+    }
+
+    return uri;
+};
+
+function build_default_whip_whep_url(query, apiPath) {
+    // The format for query string to overwrite configs of server.
+    console.log('?eip=x.x.x.x to overwrite candidate. 覆盖服务器candidate(外网IP)配置');
+    console.log('?api=x to overwrite WebRTC API(1985).');
+    console.log('?schema=http|https to overwrite WebRTC API protocol.');
+    console.log(`?path=xxx to overwrite default ${apiPath}`);
+
+    var server = (!query.server)? window.location.hostname:query.server;
+    var vhost = (!query.vhost)? window.location.hostname:query.vhost;
+    var app = (!query.app)? "live":query.app;
+    var stream = (!query.stream)? "livestream":query.stream;
+    var api = ':' + (query.api || (window.location.protocol === 'http:' ? '1985' : '1990'));
+    const realApiPath = query.path || apiPath;
+
+    var queries = [];
+    if (server !== vhost && vhost !== "__defaultVhost__") {
+        queries.push("vhost=" + vhost);
+    }
+    if (query.schema && window.location.protocol !== query.schema + ':') {
+        queries.push('schema=' + query.schema);
+    }
+    queries = user_extra_params(query, queries, true);
+
+    var uri = window.location.protocol + "//" + server + api + realApiPath + "?app=" + app + "&stream=" + stream + "&" + queries.join('&');
+    while (uri.lastIndexOf("?") === uri.length - 1) {
+        uri = uri.slice(0, uri.length - 1);
+    }
+    while (uri.lastIndexOf("&") === uri.length - 1) {
+        uri = uri.slice(0, uri.length - 1);
+    }
+
+    return uri;
+}
+
+/**
+* initialize the page.
+* @param flv_url the div id contains the flv stream url to play
+* @param hls_url the div id contains the hls stream url to play
+* @param modal_player the div id contains the modal player
+*/
+function srs_init_flv(flv_url, modal_player) {
+    update_nav();
+    if (flv_url) {
+        $(flv_url).val(build_default_flv_url());
+    }
+    if (modal_player) {
+        $(modal_player).width(srs_get_player_modal() + "px");
+        $(modal_player).css("margin-left", "-" + srs_get_player_modal() / 2 +"px");
+    }
+}
+function srs_init_rtc(id, query) {
+    update_nav();
+    $(id).val(build_default_rtc_url(query));
+}
+function srs_init_whip(id, query) {
+    update_nav();
+    $(id).val(build_default_whip_whep_url(query, '/rtc/v1/whip/'));
+}
+function srs_init_whep(id, query) {
+    update_nav();
+    $(id).val(build_default_whip_whep_url(query, '/rtc/v1/whep/'));
+}

web/html/js/srs.player.js

@@ -0,0 +1,382 @@
+/**
+* the SrsPlayer object.
+* @param container the html container id.
+* @param width a float value specifies the width of player.
+* @param height a float value specifies the height of player.
+* @param private_object [optional] an object that used as private object, 
+*       for example, the logic chat object which owner this player.
+* Usage:
+        <script type="text/javascript" src="js/swfobject.js"></script>
+        <script type="text/javascript" src="js/srs.player.js"></script>
+        <div id="player"></div>
+        var p = new SrsPlayer("player", 640, 480);
+        p.set_srs_player_url("srs_player.swf?v=1.0.0");
+        p.on_player_ready = function() {
+            p.set_bt(0.8);
+            p.set_mbt(1.2);
+            p.play("rtmp://ossrs.net/live/livestream");
+        };
+        p.on_player_metadata = function(metadata) {
+            console.log(metadata);
+            console.log(p.dump_log());
+        };
+        p.start();
+*/
+function SrsPlayer(container, width, height, private_object) {
+    if (!SrsPlayer.__id) {
+        SrsPlayer.__id = 100;
+    }
+    if (!SrsPlayer.__players) {
+        SrsPlayer.__players = [];
+    }
+    
+    SrsPlayer.__players.push(this);
+    
+    this.private_object = private_object;
+    this.container = container;
+    this.width = width;
+    this.height = height;
+    this.id = SrsPlayer.__id++;
+    this.stream_url = null;
+    this.buffer_time = 0.3; // default to 0.3
+    this.max_buffer_time = this.buffer_time * 3; // default to 3 x bufferTime.
+    this.volume = 1.0; // default to 100%
+    this.callbackObj = null;
+    this.srs_player_url = "srs_player/release/srs_player.swf?_version="+srs_get_version_code();
+    
+    // callback set the following values.
+    this.meatadata = {}; // for on_player_metadata
+    this.time = 0; // current stream time.
+    this.buffer_length = 0; // current stream buffer length.
+    this.kbps = 0; // current stream bitrate(video+audio) in kbps.
+    this.fps = 0; // current stream video fps.
+    this.rtime = 0; // flash relative time in ms.
+
+    this.__fluency = {
+        total_empty_count: 0,
+        total_empty_time: 0,
+        current_empty_time: 0
+    };
+    this.__fluency.on_stream_empty = function(time) {
+        this.total_empty_count++;
+        this.current_empty_time = time;
+    };
+    this.__fluency.on_stream_full = function(time) {
+        if (this.current_empty_time > 0) {
+            this.total_empty_time += time - this.current_empty_time;
+            this.current_empty_time = 0;
+        }
+    };
+    this.__fluency.calc = function(time) {
+        var den = this.total_empty_count * 4 + this.total_empty_time * 2 + time;
+        if (den > 0) {
+            return time * 100 / den;
+        }
+        return 0;
+    };
+}
+/**
+* user can set some callback, then start the player.
+* @param url the default url.
+* callbacks:
+*      on_player_ready():int, when srs player ready, user can play().
+*      on_player_metadata(metadata:Object):int, when srs player get metadata.
+* methods:
+*      set_bt(t:Number):void, set the buffer time in seconds.
+*      set_mbt(t:Number):void, set the max buffer time in seconds.
+*      dump_log():String, get all logs of player.
+*/
+SrsPlayer.prototype.start = function(url) {
+    if (url) {
+        this.stream_url = url;
+    }
+    
+    // embed the flash.
+    var flashvars = {};
+    flashvars.id = this.id;
+    flashvars.on_player_ready = "__srs_on_player_ready";
+    flashvars.on_player_metadata = "__srs_on_player_metadata";
+    flashvars.on_player_timer = "__srs_on_player_timer";
+    flashvars.on_player_empty = "__srs_on_player_empty";
+    flashvars.on_player_full = "__srs_on_player_full";
+    flashvars.on_player_status = "__srs_on_player_status";
+    
+    var params = {};
+    params.wmode = "opaque";
+    params.allowFullScreen = "true";
+    params.allowScriptAccess = "always";
+    
+    var attributes = {};
+    
+    var self = this;
+    
+    swfobject.embedSWF(
+        this.srs_player_url, 
+        this.container,
+        this.width, this.height,
+        "11.1.0", "js/AdobeFlashPlayerInstall.swf",
+        flashvars, params, attributes,
+        function(callbackObj){
+            self.callbackObj = callbackObj;
+            if (!callbackObj.success) {
+                console.error('Initialize player failed:'); console.error(callbackObj);
+            }
+        }
+    );
+    
+    return this;
+}
+/**
+* play the stream.
+* @param stream_url the url of stream, rtmp or http.
+* @param volume the volume, 0 is mute, 1 is 100%, 2 is 200%.
+*/
+SrsPlayer.prototype.play = function(url, volume) {
+    this.stop();
+    SrsPlayer.__players.push(this);
+    
+    if (url) {
+        this.stream_url = url;
+    }
+    
+    // volume maybe 0, so never use if(volume) to check its value.
+    if (volume != null && volume != undefined) {
+        this.volume = volume;
+    }
+    
+    this.callbackObj.ref.__play(this.stream_url, this.width, this.height, this.buffer_time, this.max_buffer_time, this.volume);
+}
+/**
+ * stop play stream.
+ */
+SrsPlayer.prototype.stop = function() {
+    this.callbackObj.ref.__stop();
+}
+/**
+ * pause the play.
+ */
+SrsPlayer.prototype.pause = function() {
+    this.callbackObj.ref.__pause();
+}
+/**
+ * resume the play.
+ */
+SrsPlayer.prototype.resume = function() {
+    this.callbackObj.ref.__resume();
+}
+/**
+ * get the stream fluency, where 100 is 100%.
+ */
+SrsPlayer.prototype.fluency = function() {
+    return this.__fluency.calc(this.rtime);
+}
+/**
+ * get the stream empty count.
+ */
+SrsPlayer.prototype.empty_count = function() {
+    return this.__fluency.total_empty_count;
+}
+/**
+ * get all log data.
+ */
+SrsPlayer.prototype.dump_log = function() {
+    return this.callbackObj.ref.__dump_log();
+}
+/**
+* to set the DAR, for example, DAR=16:9 where num=16,den=9.
+* @param num, for example, 16. 
+*       use metadata width if 0.
+*       use user specified width if -1.
+* @param den, for example, 9. 
+*       use metadata height if 0.
+*       use user specified height if -1.
+*/
+SrsPlayer.prototype.set_dar = function(num, den) {
+    this.callbackObj.ref.__set_dar(num, den);
+}
+/**
+* set the fullscreen size data.
+* @refer the refer fullscreen mode. it can be:
+*       video: use video orignal size.
+*       screen: use screen size to rescale video.
+* @param percent, the rescale percent, where
+*       100 means 100%.
+*/
+SrsPlayer.prototype.set_fs = function(refer, percent) {
+    this.callbackObj.ref.__set_fs(refer, percent);
+}
+/**
+* set the stream buffer time in seconds.
+* @buffer_time the buffer time in seconds.
+*/
+SrsPlayer.prototype.set_bt = function(buffer_time) {
+    if (this.buffer_time == buffer_time) {
+        return;
+    }
+
+    this.buffer_time = buffer_time;
+    this.callbackObj.ref.__set_bt(buffer_time);
+
+    // reset the max buffer time to 3 x buffer_time.
+    this.set_mbt(buffer_time * 3);
+}
+/**
+ * set the stream max buffer time in seconds.
+ * @param max_buffer_time the max buffer time in seconds.
+ * @remark this is the key feature for realtime communication by flash.
+ */
+SrsPlayer.prototype.set_mbt = function(max_buffer_time) {
+    // we must atleast set the max buffer time to 0.6s.
+    max_buffer_time = Math.max(0.6, max_buffer_time);
+    // max buffer time always greater than buffer time.
+    max_buffer_time = Math.max(this.buffer_time, max_buffer_time);
+
+    if (parseInt(this.max_buffer_time * 10) == parseInt(max_buffer_time * 10)) {
+        return;
+    }
+
+    this.max_buffer_time = max_buffer_time;
+    this.callbackObj.ref.__set_mbt(max_buffer_time);
+}
+/**
+ * set the srs_player.swf url
+ * @param url, srs_player.swf's url.
+ * @param params, object.
+ */
+ SrsPlayer.prototype.set_srs_player_url = function(url, params) {
+    var query_array = [], 
+        query_string = "", 
+        p;
+    params = params || {}; 
+    params._version = srs_get_version_code();
+    for (p in params) {
+        if (params.hasOwnProperty(p)) {
+            query_array.push(p + "=" + encodeURIComponent(params[p]));
+        }
+    }   
+    query_string = query_array.join("&");
+    this.srs_player_url = url + "?" + query_string;
+}
+ /**
+  * the callback when player is ready.
+  */
+SrsPlayer.prototype.on_player_ready = function() {
+}
+/**
+ * the callback when player got metadata.
+ * @param metadata the metadata which player got.
+ */
+SrsPlayer.prototype.on_player_metadata = function(metadata) {
+    // ignore.
+}
+/**
+ * the callback when player timer event.
+ * @param time current stream time.
+ * @param buffer_length current buffer length.
+ * @param kbps current video plus audio bitrate in kbps.
+ * @param fps current video fps.
+ * @param rtime current relative time by flash.util.getTimer().
+ */
+SrsPlayer.prototype.on_player_timer = function(time, buffer_length, kbps, fps, rtime) {
+    // ignore.
+}
+/**
+ * the callback when player got NetStream.Buffer.Empty
+ * @param time current relative time by flash.util.getTimer().
+ */
+SrsPlayer.prototype.on_player_empty = function(time) {
+    // ignore.
+}
+/**
+ * the callback when player got NetStream.Buffer.Full
+ * @param time current relative time by flash.util.getTimer().
+ */
+SrsPlayer.prototype.on_player_full = function(time) {
+    // ignore.
+}
+/**
+ * the callback when player status change.
+ * @param code the status code, "init", "connected", "play", "closed", "rejected", "failed".
+ *      init => connected/rejected/failed
+ *      connected => play/rejected => closed
+ * @param desc the description for the status.
+ */
+SrsPlayer.prototype.on_player_status = function(code, desc) {
+    // ignore.
+}
+
+/**
+ * helpers.
+ */
+function __srs_find_player(id) {
+    for (var i = 0; i < SrsPlayer.__players.length; i++) {
+        var player = SrsPlayer.__players[i];
+        
+        if (player.id != id) {
+            continue;
+        }
+        
+        return player;
+    }
+    
+    throw new Error("player not found. id=" + id);
+}
+function __srs_on_player_ready(id) {
+    var player = __srs_find_player(id);
+    player.on_player_ready();
+}
+function __srs_on_player_metadata(id, metadata) {
+    var player = __srs_find_player(id);
+    
+    // user may override the on_player_metadata, 
+    // so set the data before invoke it.
+    player.metadata = metadata;
+    
+    player.on_player_metadata(metadata);
+}
+function __srs_on_player_timer(id, time, buffer_length, kbps, fps, rtime) {
+    var player = __srs_find_player(id);
+    
+    buffer_length = Math.max(0, buffer_length);
+    buffer_length = Math.min(player.buffer_time, buffer_length);
+    
+    time = Math.max(0, time);
+    
+    // user may override the on_player_timer, 
+    // so set the data before invoke it.
+    player.time = time;
+    player.buffer_length = buffer_length;
+    player.kbps = kbps;
+    player.fps = fps;
+    player.rtime = rtime;
+
+    player.on_player_timer(time, buffer_length, kbps, fps, rtime);
+}
+function __srs_on_player_empty(id, time) {
+    var player = __srs_find_player(id);
+    player.__fluency.on_stream_empty(time);
+    player.on_player_empty(time);
+}
+function __srs_on_player_full(id, time) {
+    var player = __srs_find_player(id);
+    player.__fluency.on_stream_full(time);
+    player.on_player_full(time);
+}
+function __srs_on_player_status(id, code, desc) {
+    var player = __srs_find_player(id);
+    player.on_player_status(code, desc);
+
+    if (code != "closed") {
+        return;
+    }
+    for (var i = 0; i < SrsPlayer.__players.length; i++) {
+        var player = SrsPlayer.__players[i];
+
+        if (player.id != this.id) {
+            continue;
+        }
+
+        SrsPlayer.__players.splice(i, 1);
+        break;
+    }
+}

web/html/js/srs.publisher.js

@@ -0,0 +1,173 @@
+/**
+* the SrsPublisher object.
+* @param container the html container id.
+* @param width a float value specifies the width of publisher.
+* @param height a float value specifies the height of publisher.
+* @param private_object [optional] an object that used as private object, 
+*       for example, the logic chat object which owner this publisher.
+*/
+function SrsPublisher(container, width, height, private_object) {
+    if (!SrsPublisher.__id) {
+        SrsPublisher.__id = 100;
+    }
+    if (!SrsPublisher.__publishers) {
+        SrsPublisher.__publishers = [];
+    }
+    
+    SrsPublisher.__publishers.push(this);
+    
+    this.private_object = private_object;
+    this.container = container;
+    this.width = width;
+    this.height = height;
+    this.id = SrsPublisher.__id++;
+    this.callbackObj = null;
+    
+    // set the values when publish.
+    this.url = null;
+    this.vcodec = {};
+    this.acodec = {};
+    
+    // callback set the following values.
+    this.cameras = [];
+    this.microphones = [];
+    this.code = 0;
+    
+    // error code defines.
+    this.errors = {
+        "100": "无法获取指定的摄像头。", //error_camera_get 
+        "101": "无法获取指定的麦克风。", //error_microphone_get 
+        "102": "摄像头为禁用状态，推流时请允许flash访问摄像头。", //error_camera_muted 
+        "103": "服务器关闭了连接。", //error_connection_closed 
+        "104": "服务器连接失败。", //error_connection_failed 
+        "199": "未知错误。"
+    };
+}
+/**
+* user can set some callback, then start the publisher.
+* callbacks:
+*      on_publisher_ready(cameras, microphones):int, when srs publisher ready, user can publish.
+*      on_publisher_error(code, desc):int, when srs publisher error, callback this method.
+*      on_publisher_warn(code, desc):int, when srs publisher warn, callback this method.
+*/
+SrsPublisher.prototype.start = function() {
+    // embed the flash.
+    var flashvars = {};
+    flashvars.id = this.id;
+    flashvars.width = this.width;
+    flashvars.height = this.height;
+    flashvars.on_publisher_ready = "__srs_on_publisher_ready";
+    flashvars.on_publisher_error = "__srs_on_publisher_error";
+    flashvars.on_publisher_warn = "__srs_on_publisher_warn";
+    
+    var params = {};
+    params.wmode = "opaque";
+    params.allowFullScreen = "true";
+    params.allowScriptAccess = "always";
+    
+    var attributes = {};
+    
+    var self = this;
+    
+    swfobject.embedSWF(
+        "srs_publisher/release/srs_publisher.swf?_version="+srs_get_version_code(), 
+        this.container,
+        this.width, this.height,
+        "11.1.0", "js/AdobeFlashPlayerInstall.swf",
+        flashvars, params, attributes,
+        function(callbackObj){
+            self.callbackObj = callbackObj;
+        }
+    );
+    
+    return this;
+}
+/**
+* publish stream to server.
+* @param url a string indicates the rtmp url to publish.
+* @param vcodec an object contains the video codec info.
+* @param acodec an object contains the audio codec info.
+*/
+SrsPublisher.prototype.publish = function(url, vcodec, acodec) {
+    this.stop();
+    SrsPublisher.__publishers.push(this);
+    
+    if (url) {
+        this.url = url;
+    }
+    if (vcodec) {
+        this.vcodec = vcodec;
+    }
+    if (acodec) {
+        this.acodec = acodec;
+    }
+    
+    this.callbackObj.ref.__publish(this.url, this.width, this.height, this.vcodec, this.acodec);
+}
+SrsPublisher.prototype.stop = function() {
+    for (var i = 0; i < SrsPublisher.__publishers.length; i++) {
+        var player = SrsPublisher.__publishers[i];
+        
+        if (player.id != this.id) {
+            continue;
+        }
+        
+        SrsPublisher.__publishers.splice(i, 1);
+        break;
+    }
+    
+    this.callbackObj.ref.__stop();
+}
+/**
+* when publisher ready.
+* @param cameras a string array contains the names of cameras.
+* @param microphones a string array contains the names of microphones.
+*/
+SrsPublisher.prototype.on_publisher_ready = function(cameras, microphones) {
+}
+/**
+* when publisher error.
+* @code the error code.
+* @desc the error desc message.
+*/
+SrsPublisher.prototype.on_publisher_error = function(code, desc) {
+    throw new Error("publisher error. code=" + code + ", desc=" + desc);
+}
+SrsPublisher.prototype.on_publisher_warn = function(code, desc) {
+    throw new Error("publisher warn. code=" + code + ", desc=" + desc);
+}
+function __srs_find_publisher(id) {
+    for (var i = 0; i < SrsPublisher.__publishers.length; i++) {
+        var publisher = SrsPublisher.__publishers[i];
+        
+        if (publisher.id != id) {
+            continue;
+        }
+        
+        return publisher;
+    }
+    
+    throw new Error("publisher not found. id=" + id);
+}
+function __srs_on_publisher_ready(id, cameras, microphones) {
+    var publisher = __srs_find_publisher(id);
+    
+    publisher.cameras = cameras;
+    publisher.microphones = microphones;
+    
+    publisher.on_publisher_ready(cameras, microphones);
+}
+function __srs_on_publisher_error(id, code) {
+    var publisher = __srs_find_publisher(id);
+    
+    publisher.code = code;
+    
+    publisher.on_publisher_error(code, publisher.errors[""+code]);
+}
+function __srs_on_publisher_warn(id, code) {
+    var publisher = __srs_find_publisher(id);
+    
+    publisher.code = code;
+    
+    publisher.on_publisher_warn(code, publisher.errors[""+code]);
+}

web/html/js/srs.sdk.js

@@ -0,0 +1,681 @@
+
+//
+// Copyright (c) 2013-2021 Winlin
+//
+// SPDX-License-Identifier: MIT
+//
+
+'use strict';
+
+function SrsError(name, message) {
+    this.name = name;
+    this.message = message;
+    this.stack = (new Error()).stack;
+}
+SrsError.prototype = Object.create(Error.prototype);
+SrsError.prototype.constructor = SrsError;
+
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-awat-prmise based SRS RTC Publisher.
+function SrsRtcPublisherAsync() {
+    var self = {};
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+    self.constraints = {
+        audio: true,
+        video: {
+            width: {ideal: 320, max: 576}
+        }
+    };
+
+    // @see https://github.com/rtcdn/rtcdn-draft
+    // @url The WebRTC url to play with, for example:
+    //      webrtc://r.ossrs.net/live/livestream
+    // or specifies the API port:
+    //      webrtc://r.ossrs.net:11985/live/livestream
+    // or autostart the publish:
+    //      webrtc://r.ossrs.net/live/livestream?autostart=true
+    // or change the app from live to myapp:
+    //      webrtc://r.ossrs.net:11985/myapp/livestream
+    // or change the stream from livestream to mystream:
+    //      webrtc://r.ossrs.net:11985/live/mystream
+    // or set the api server to myapi.domain.com:
+    //      webrtc://myapi.domain.com/live/livestream
+    // or set the candidate(eip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
+    // or force to access https API:
+    //      webrtc://r.ossrs.net/live/livestream?schema=https
+    // or use plaintext, without SRTP:
+    //      webrtc://r.ossrs.net/live/livestream?encrypt=false
+    // or any other information, will pass-by in the query:
+    //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
+    //      webrtc://r.ossrs.net/live/livestream?token=xxx
+    self.publish = async function (url) {
+        var conf = self.__internal.prepareUrl(url);
+        self.pc.addTransceiver("audio", {direction: "sendonly"});
+        self.pc.addTransceiver("video", {direction: "sendonly"});
+        //self.pc.addTransceiver("video", {direction: "sendonly"});
+        //self.pc.addTransceiver("audio", {direction: "sendonly"});
+
+        if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
+            throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
+        }
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+        stream.getTracks().forEach(function (track) {
+            self.pc.addTrack(track);
+
+            // Notify about local track when stream is ok.
+            self.ontrack && self.ontrack({track: track});
+        });
+
+        var offer = await self.pc.createOffer();
+        await self.pc.setLocalDescription(offer);
+        var session = await new Promise(function (resolve, reject) {
+            // @see https://github.com/rtcdn/rtcdn-draft
+            var data = {
+                api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
+                clientip: null, sdp: offer.sdp
+            };
+            console.log("Generated offer: ", data);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+                const data = JSON.parse(xhr.responseText);
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', conf.apiUrl, true);
+            xhr.setRequestHeader('Content-type', 'application/json');
+            xhr.send(JSON.stringify(data));
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+        );
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+        return session;
+    };
+
+    // Close the publisher.
+    self.close = function () {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got local stream.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    self.ontrack = function (event) {
+        // Add track to stream of SDK.
+        self.stream.addTrack(event.track);
+    };
+
+    // Internal APIs.
+    self.__internal = {
+        defaultPath: '/rtc/v1/publish/',
+        prepareUrl: function (webrtcUrl) {
+            var urlObject = self.__internal.parse(webrtcUrl);
+
+            // If user specifies the schema, use it as API schema.
+            var schema = urlObject.user_query.schema;
+            schema = schema ? schema + ':' : window.location.protocol;
+
+            var port = urlObject.port || 1985;
+            if (schema === 'https:') {
+                port = urlObject.port || 443;
+            }
+
+            // @see https://github.com/rtcdn/rtcdn-draft
+            var api = urlObject.user_query.play || self.__internal.defaultPath;
+            if (api.lastIndexOf('/') !== api.length - 1) {
+                api += '/';
+            }
+
+            var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+            for (var key in urlObject.user_query) {
+                if (key !== 'api' && key !== 'play') {
+                    apiUrl += '&' + key + '=' + urlObject.user_query[key];
+                }
+            }
+            // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
+            apiUrl = apiUrl.replace(api + '&', api + '?');
+
+            var streamUrl = urlObject.url;
+
+            return {
+                apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
+                tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
+            };
+        },
+        parse: function (url) {
+            // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
+            var a = document.createElement("a");
+            a.href = url.replace("rtmp://", "http://")
+                .replace("webrtc://", "http://")
+                .replace("rtc://", "http://");
+
+            var vhost = a.hostname;
+            var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
+            var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
+
+            // parse the vhost in the params of app, that srs supports.
+            app = app.replace("...vhost...", "?vhost=");
+            if (app.indexOf("?") >= 0) {
+                var params = app.slice(app.indexOf("?"));
+                app = app.slice(0, app.indexOf("?"));
+
+                if (params.indexOf("vhost=") > 0) {
+                    vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+                    if (vhost.indexOf("&") > 0) {
+                        vhost = vhost.slice(0, vhost.indexOf("&"));
+                    }
+                }
+            }
+
+            // when vhost equals to server, and server is ip,
+            // the vhost is __defaultVhost__
+            if (a.hostname === vhost) {
+                var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
+                if (re.test(a.hostname)) {
+                    vhost = "__defaultVhost__";
+                }
+            }
+
+            // parse the schema
+            var schema = "rtmp";
+            if (url.indexOf("://") > 0) {
+                schema = url.slice(0, url.indexOf("://"));
+            }
+
+            var port = a.port;
+            if (!port) {
+                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
+                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
+                    port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
+                }
+
+                // Guess by schema.
+                if (schema === 'http') {
+                    port = 80;
+                } else if (schema === 'https') {
+                    port = 443;
+                } else if (schema === 'rtmp') {
+                    port = 1935;
+                }
+            }
+
+            var ret = {
+                url: url,
+                schema: schema,
+                server: a.hostname, port: port,
+                vhost: vhost, app: app, stream: stream
+            };
+            self.__internal.fill_query(a.search, ret);
+
+            // For webrtc API, we use 443 if page is https, or schema specified it.
+            if (!ret.port) {
+                if (schema === 'webrtc' || schema === 'rtc') {
+                    if (ret.user_query.schema === 'https') {
+                        ret.port = 443;
+                    } else if (window.location.href.indexOf('https://') === 0) {
+                        ret.port = 443;
+                    } else {
+                        // For WebRTC, SRS use 1985 as default API port.
+                        ret.port = 1985;
+                    }
+                }
+            }
+
+            return ret;
+        },
+        fill_query: function (query_string, obj) {
+            // pure user query object.
+            obj.user_query = {};
+
+            if (query_string.length === 0) {
+                return;
+            }
+
+            // split again for angularjs.
+            if (query_string.indexOf("?") >= 0) {
+                query_string = query_string.split("?")[1];
+            }
+
+            var queries = query_string.split("&");
+            for (var i = 0; i < queries.length; i++) {
+                var elem = queries[i];
+
+                var query = elem.split("=");
+                obj[query[0]] = query[1];
+                obj.user_query[query[0]] = query[1];
+            }
+
+            // alias domain for vhost.
+            if (obj.domain) {
+                obj.vhost = obj.domain;
+            }
+        }
+    };
+
+    self.pc = new RTCPeerConnection(null);
+
+    // To keep api consistent between player and publisher.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    // @see https://webrtc.org/getting-started/media-devices
+    self.stream = new MediaStream();
+
+    return self;
+}
+
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-await-promise based SRS RTC Player.
+function SrsRtcPlayerAsync() {
+    var self = {};
+
+    // @see https://github.com/rtcdn/rtcdn-draft
+    // @url The WebRTC url to play with, for example:
+    //      webrtc://r.ossrs.net/live/livestream
+    // or specifies the API port:
+    //      webrtc://r.ossrs.net:11985/live/livestream
+    //      webrtc://r.ossrs.net:80/live/livestream
+    // or autostart the play:
+    //      webrtc://r.ossrs.net/live/livestream?autostart=true
+    // or change the app from live to myapp:
+    //      webrtc://r.ossrs.net:11985/myapp/livestream
+    // or change the stream from livestream to mystream:
+    //      webrtc://r.ossrs.net:11985/live/mystream
+    // or set the api server to myapi.domain.com:
+    //      webrtc://myapi.domain.com/live/livestream
+    // or set the candidate(eip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
+    // or force to access https API:
+    //      webrtc://r.ossrs.net/live/livestream?schema=https
+    // or use plaintext, without SRTP:
+    //      webrtc://r.ossrs.net/live/livestream?encrypt=false
+    // or any other information, will pass-by in the query:
+    //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
+    //      webrtc://r.ossrs.net/live/livestream?token=xxx
+    self.play = async function(url) {
+        var conf = self.__internal.prepareUrl(url);
+        self.pc.addTransceiver("audio", {direction: "recvonly"});
+        self.pc.addTransceiver("video", {direction: "recvonly"});
+        //self.pc.addTransceiver("video", {direction: "recvonly"});
+        //self.pc.addTransceiver("audio", {direction: "recvonly"});
+
+        var offer = await self.pc.createOffer();
+        await self.pc.setLocalDescription(offer);
+        var session = await new Promise(function(resolve, reject) {
+            // @see https://github.com/rtcdn/rtcdn-draft
+            var data = {
+                api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
+                clientip: null, sdp: offer.sdp
+            };
+            console.log("Generated offer: ", data);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+                const data = JSON.parse(xhr.responseText);
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', conf.apiUrl, true);
+            xhr.setRequestHeader('Content-type', 'application/json');
+            xhr.send(JSON.stringify(data));
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+        );
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+        return session;
+    };
+
+    // Close the player.
+    self.close = function() {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got remote track.
+    // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
+    self.ontrack = function (event) {
+        // https://webrtc.org/getting-started/remote-streams
+        self.stream.addTrack(event.track);
+    };
+
+    // Internal APIs.
+    self.__internal = {
+        defaultPath: '/rtc/v1/play/',
+        prepareUrl: function (webrtcUrl) {
+            var urlObject = self.__internal.parse(webrtcUrl);
+
+            // If user specifies the schema, use it as API schema.
+            var schema = urlObject.user_query.schema;
+            schema = schema ? schema + ':' : window.location.protocol;
+
+            var port = urlObject.port || 1985;
+            if (schema === 'https:') {
+                port = urlObject.port || 443;
+            }
+
+            // @see https://github.com/rtcdn/rtcdn-draft
+            var api = urlObject.user_query.play || self.__internal.defaultPath;
+            if (api.lastIndexOf('/') !== api.length - 1) {
+                api += '/';
+            }
+
+            var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+            for (var key in urlObject.user_query) {
+                if (key !== 'api' && key !== 'play') {
+                    apiUrl += '&' + key + '=' + urlObject.user_query[key];
+                }
+            }
+            // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
+            apiUrl = apiUrl.replace(api + '&', api + '?');
+
+            var streamUrl = urlObject.url;
+
+            return {
+                apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
+                tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
+            };
+        },
+        parse: function (url) {
+            // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
+            var a = document.createElement("a");
+            a.href = url.replace("rtmp://", "http://")
+                .replace("webrtc://", "http://")
+                .replace("rtc://", "http://");
+
+            var vhost = a.hostname;
+            var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
+            var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
+
+            // parse the vhost in the params of app, that srs supports.
+            app = app.replace("...vhost...", "?vhost=");
+            if (app.indexOf("?") >= 0) {
+                var params = app.slice(app.indexOf("?"));
+                app = app.slice(0, app.indexOf("?"));
+
+                if (params.indexOf("vhost=") > 0) {
+                    vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+                    if (vhost.indexOf("&") > 0) {
+                        vhost = vhost.slice(0, vhost.indexOf("&"));
+                    }
+                }
+            }
+
+            // when vhost equals to server, and server is ip,
+            // the vhost is __defaultVhost__
+            if (a.hostname === vhost) {
+                var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
+                if (re.test(a.hostname)) {
+                    vhost = "__defaultVhost__";
+                }
+            }
+
+            // parse the schema
+            var schema = "rtmp";
+            if (url.indexOf("://") > 0) {
+                schema = url.slice(0, url.indexOf("://"));
+            }
+
+            var port = a.port;
+            if (!port) {
+                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
+                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
+                    port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
+                }
+
+                // Guess by schema.
+                if (schema === 'http') {
+                    port = 80;
+                } else if (schema === 'https') {
+                    port = 443;
+                } else if (schema === 'rtmp') {
+                    port = 1935;
+                }
+            }
+
+            var ret = {
+                url: url,
+                schema: schema,
+                server: a.hostname, port: port,
+                vhost: vhost, app: app, stream: stream
+            };
+            self.__internal.fill_query(a.search, ret);
+
+            // For webrtc API, we use 443 if page is https, or schema specified it.
+            if (!ret.port) {
+                if (schema === 'webrtc' || schema === 'rtc') {
+                    if (ret.user_query.schema === 'https') {
+                        ret.port = 443;
+                    } else if (window.location.href.indexOf('https://') === 0) {
+                        ret.port = 443;
+                    } else {
+                        // For WebRTC, SRS use 1985 as default API port.
+                        ret.port = 1985;
+                    }
+                }
+            }
+
+            return ret;
+        },
+        fill_query: function (query_string, obj) {
+            // pure user query object.
+            obj.user_query = {};
+
+            if (query_string.length === 0) {
+                return;
+            }
+
+            // split again for angularjs.
+            if (query_string.indexOf("?") >= 0) {
+                query_string = query_string.split("?")[1];
+            }
+
+            var queries = query_string.split("&");
+            for (var i = 0; i < queries.length; i++) {
+                var elem = queries[i];
+
+                var query = elem.split("=");
+                obj[query[0]] = query[1];
+                obj.user_query[query[0]] = query[1];
+            }
+
+            // alias domain for vhost.
+            if (obj.domain) {
+                obj.vhost = obj.domain;
+            }
+        }
+    };
+
+    self.pc = new RTCPeerConnection(null);
+
+    // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
+    self.stream = new MediaStream();
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
+    self.pc.ontrack = function(event) {
+        if (self.ontrack) {
+            self.ontrack(event);
+        }
+    };
+
+    return self;
+}
+
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-awat-prmise based SRS RTC Publisher by WHIP.
+function SrsRtcWhipWhepAsync() {
+    var self = {};
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+    self.constraints = {
+        audio: true,
+        video: {
+            width: {ideal: 320, max: 576}
+        }
+    };
+
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+    // @url The WebRTC url to publish with, for example:
+    //      http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
+    self.publish = async function (url) {
+        if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
+
+        self.pc.addTransceiver("audio", {direction: "sendonly"});
+        self.pc.addTransceiver("video", {direction: "sendonly"});
+
+        if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
+            throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
+        }
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+        stream.getTracks().forEach(function (track) {
+            self.pc.addTrack(track);
+
+            // Notify about local track when stream is ok.
+            self.ontrack && self.ontrack({track: track});
+        });
+
+        var offer = await self.pc.createOffer();
+        await self.pc.setLocalDescription(offer);
+        const answer = await new Promise(function (resolve, reject) {
+            console.log("Generated offer: ", offer);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+                const data = xhr.responseText;
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', url, true);
+            xhr.setRequestHeader('Content-type', 'application/sdp');
+            xhr.send(offer.sdp);
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: answer})
+        );
+
+        return self.__internal.parseId(url, offer.sdp, answer);
+    };
+
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+    // @url The WebRTC url to play with, for example:
+    //      http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
+    self.play = async function(url) {
+        if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
+
+        self.pc.addTransceiver("audio", {direction: "recvonly"});
+        self.pc.addTransceiver("video", {direction: "recvonly"});
+
+        var offer = await self.pc.createOffer();
+        await self.pc.setLocalDescription(offer);
+        const answer = await new Promise(function(resolve, reject) {
+            console.log("Generated offer: ", offer);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+                const data = xhr.responseText;
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', url, true);
+            xhr.setRequestHeader('Content-type', 'application/sdp');
+            xhr.send(offer.sdp);
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: answer})
+        );
+
+        return self.__internal.parseId(url, offer.sdp, answer);
+    };
+
+    // Close the publisher.
+    self.close = function () {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got local stream.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    self.ontrack = function (event) {
+        // Add track to stream of SDK.
+        self.stream.addTrack(event.track);
+    };
+
+    self.pc = new RTCPeerConnection(null);
+
+    // To keep api consistent between player and publisher.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    // @see https://webrtc.org/getting-started/media-devices
+    self.stream = new MediaStream();
+
+    // Internal APIs.
+    self.__internal = {
+        parseId: (url, offer, answer) => {
+            let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
+            sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
+            sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
+            sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
+
+            const a = document.createElement("a");
+            a.href = url;
+            return {
+                sessionid: sessionid, // Should be ice-ufrag of answer:offer.
+                simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
+            };
+        },
+    };
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
+    self.pc.ontrack = function(event) {
+        if (self.ontrack) {
+            self.ontrack(event);
+        }
+    };
+
+    return self;
+}
+
+// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
+// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
+function SrsRtcFormatSenders(senders, kind) {
+    var codecs = [];
+    senders.forEach(function (sender) {
+        var params = sender.getParameters();
+        params && params.codecs && params.codecs.forEach(function(c) {
+            if (kind && sender.track.kind !== kind) {
+                return;
+            }
+
+            if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
+                return;
+            }
+
+            var s = '';
+
+            s += c.mimeType.replace('audio/', '').replace('video/', '');
+            s += ', ' + c.clockRate + 'HZ';
+            if (sender.track.kind === "audio") {
+                s += ', channels: ' + c.channels;
+            }
+            s += ', pt: ' + c.payloadType;
+
+            codecs.push(s);
+        });
+    });
+    return codecs.join(", ");
+}
+

web/html/js/srs.utility.js

@@ -0,0 +1,8 @@
+/**
+* parse the rtmp url,
+* for example: rtmp://demo.srs.com:1935/live...vhost...players/livestream
+* @return object {server, port, vhost, app, stream}
+*/
+function srs_parse_rtmp_url(rtmp_url) {
+    return parse_rtmp_url(rtmp_url);
+}

web/html/js/winlin.utility.js

@@ -0,0 +1,686 @@
+// winlin.utility.js
+
+/**
+ * common utilities
+ * depends: jquery1.10
+ * https://gitee.com/winlinvip/codes/rpn0c2ewbomj81augzk4y59
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * v 1.0.23
+ */
+
+/**
+ * padding the output.
+ * padding(3, 5, '0') is 00003
+ * padding(3, 5, 'x') is xxxx3
+ * @see http://blog.csdn.net/win_lin/article/details/12065413
+ */
+function padding(number, length, prefix) {
+    if(String(number).length >= length){
+        return String(number);
+    }
+    return padding(prefix+number, length, prefix);
+}
+
+/**
+ * extends system array, to remove all specified elem.
+ * @param arr the array to remove elem from.
+ * @param elem the elem to remove.
+ * @remark all elem will be removed.
+ * for example,
+ *      arr = [10, 15, 20, 30, 20, 40]
+ *      system_array_remove(arr, 10) // arr=[15, 20, 30, 20, 40]
+ *      system_array_remove(arr, 20) // arr=[15, 30, 40]
+ */
+function system_array_remove(arr, elem) {
+    if (!arr) {
+        return;
+    }
+
+    var removed = true;
+    var i = 0;
+    while (removed) {
+        removed = false;
+        for (; i < arr.length; i++) {
+            if (elem == arr[i]) {
+                arr.splice(i, 1);
+                removed = true;
+                break;
+            }
+        }
+    }
+}
+
+/**
+ * whether the array contains specified element.
+ * @param arr the array to find.
+ * @param elem_or_function the element value or compare function.
+ * @returns true contains elem; otherwise false.
+ * for example,
+ *      arr = [10, 15, 20, 30, 20, 40]
+ *      system_array_contains(arr, 10) // true
+ *      system_array_contains(arr, 11) // false
+ *      system_array_contains(arr, function(elem){return elem == 30;}); // true
+ *      system_array_contains(arr, function(elem){return elem == 60;}); // false
+ */
+function system_array_contains(arr, elem_or_function) {
+    return system_array_get(arr, elem_or_function) != null;
+}
+
+/**
+ * get the specified element from array
+ * @param arr the array to find.
+ * @param elem_or_function the element value or compare function.
+ * @returns the matched elem; otherwise null.
+ * for example,
+ *      arr = [10, 15, 20, 30, 20, 40]
+ *      system_array_get(arr, 10) // 10
+ *      system_array_get(arr, 11) // null
+ *      system_array_get(arr, function(elem){return elem == 30;}); // 30
+ *      system_array_get(arr, function(elem){return elem == 60;}); // null
+ */
+function system_array_get(arr, elem_or_function) {
+    for (var i = 0; i < arr.length; i++) {
+        if (typeof elem_or_function == "function") {
+            if (elem_or_function(arr[i])) {
+                return arr[i];
+            }
+        } else {
+            if (elem_or_function == arr[i]) {
+                return arr[i];
+            }
+        }
+    }
+    return null;
+}
+
+/**
+ * to iterate on array.
+ * @param arr the array to iterate on.
+ * @param pfn the function to apply on it. return false to break loop.
+ * for example,
+ *      arr = [10, 15, 20, 30, 20, 40]
+ *      system_array_foreach(arr, function(elem, index){
+ *          console.log('index=' + index + ',elem=' + elem);
+ *      });
+ * @return true when iterate all elems.
+ */
+function system_array_foreach(arr, pfn) {
+    if (!pfn) {
+        return false;
+    }
+
+    for (var i = 0; i < arr.length; i++) {
+        if (!pfn(arr[i], i)) {
+            return false;
+        }
+    }
+
+    return true;
+}
+
+/**
+ * whether the str starts with flag.
+ */
+function system_string_startswith(str, flag) {
+    if (typeof flag == "object" && flag.constructor == Array) {
+        for (var i = 0; i < flag.length; i++) {
+            if (system_string_startswith(str, flag[i])) {
+                return true;
+            }
+        }
+    }
+
+    return str && flag && str.length >= flag.length && str.indexOf(flag) == 0;
+}
+
+/**
+ * whether the str ends with flag.
+ */
+function system_string_endswith(str, flag) {
+    if (typeof flag == "object" && flag.constructor == Array) {
+        for (var i = 0; i < flag.length; i++) {
+            if (system_string_endswith(str, flag[i])) {
+                return true;
+            }
+        }
+    }
+
+    return str && flag && str.length >= flag.length && str.indexOf(flag) == str.length - flag.length;
+}
+
+/**
+ * trim the start and end of flag in str.
+ * @param flag a string to trim.
+ */
+function system_string_trim(str, flag) {
+    if (!flag || !flag.length || typeof flag != "string") {
+        return str;
+    }
+
+    while (system_string_startswith(str, flag)) {
+        str = str.slice(flag.length);
+    }
+
+    while (system_string_endswith(str, flag)) {
+        str = str.slice(0, str.length - flag.length);
+    }
+
+    return str;
+}
+
+/**
+ * array sort asc, for example:
+ * [a, b] in [10, 11, 9]
+ * then sort to: [9, 10, 11]
+ * Usage, for example:
+ obj.data.data.sort(function(a, b){
+            return array_sort_asc(a.metadata.meta_id, b.metadata.meta_id);
+        });
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * @remark, if need desc, use -1*array_sort_asc(a,b)
+ */
+function array_sort_asc(elem_a, elem_b) {
+    if (elem_a > elem_b) {
+        return 1;
+    }
+    return (elem_a < elem_b)? -1 : 0;
+}
+function array_sort_desc(elem_a, elem_b) {
+    return -1 * array_sort_asc(elem_a, elem_b);
+}
+function system_array_sort_asc(elem_a, elem_b) {
+    return array_sort_asc(elem_a, elem_b);
+}
+function system_array_sort_desc(elem_a, elem_b) {
+    return -1 * array_sort_asc(elem_a, elem_b);
+}
+
+/**
+ * parse the query string to object.
+ * parse the url location object as: host(hostname:http_port), pathname(dir/filename)
+ * for example, url http://192.168.1.168:1980/ui/players.html?vhost=player.vhost.com&app=test&stream=livestream
+ * parsed to object:
+ {
+     host        : "192.168.1.168:1980",
+     hostname    : "192.168.1.168",
+     http_port   : 1980,
+     pathname    : "/ui/players.html",
+     dir         : "/ui",
+     filename    : "/players.html",
+
+     vhost       : "player.vhost.com",
+     app         : "test",
+     stream      : "livestream"
+ }
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ */
+function parse_query_string(){
+    var obj = {};
+
+    // add the uri object.
+    // parse the host(hostname:http_port), pathname(dir/filename)
+    obj.host = window.location.host;
+    obj.hostname = window.location.hostname;
+    obj.http_port = (window.location.port == "")? 80:window.location.port;
+    obj.pathname = window.location.pathname;
+    if (obj.pathname.lastIndexOf("/") <= 0) {
+        obj.dir = "/";
+        obj.filename = "";
+    } else {
+        obj.dir = obj.pathname.slice(0, obj.pathname.lastIndexOf("/"));
+        obj.filename = obj.pathname.slice(obj.pathname.lastIndexOf("/"));
+    }
+
+    // pure user query object.
+    obj.user_query = {};
+
+    // parse the query string.
+    var query_string = String(window.location.search).replace(" ", "").split("?")[1];
+    if(query_string === undefined){
+        query_string = String(window.location.hash).replace(" ", "").split("#")[1];
+        if(query_string === undefined){
+            return obj;
+        }
+    }
+
+    __fill_query(query_string, obj);
+
+    return obj;
+}
+
+function __fill_query(query_string, obj) {
+    // pure user query object.
+    obj.user_query = {};
+
+    if (query_string.length === 0) {
+        return;
+    }
+
+    // split again for angularjs.
+    if (query_string.indexOf("?") >= 0) {
+        query_string = query_string.split("?")[1];
+    }
+
+    var queries = query_string.split("&");
+    for (var i = 0; i < queries.length; i++) {
+        var elem = queries[i];
+
+        var query = elem.split("=");
+        obj[query[0]] = query[1];
+        obj.user_query[query[0]] = query[1];
+    }
+
+    // alias domain for vhost.
+    if (obj.domain) {
+        obj.vhost = obj.domain;
+    }
+}
+
+/**
+ * parse the rtmp url,
+ * for example: rtmp://demo.srs.com:1935/live...vhost...players/livestream
+ * @return object {server, port, vhost, app, stream}
+ * for exmaple, rtmp_url is rtmp://demo.srs.com:1935/live...vhost...players/livestream
+ * parsed to object:
+ {
+    server: "demo.srs.com",
+    port: 1935,
+    vhost: "players",
+    app: "live",
+    stream: "livestream"
+ }
+ */
+function parse_rtmp_url(rtmp_url) {
+    // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
+    var a = document.createElement("a");
+    a.href = rtmp_url.replace("rtmp://", "http://")
+        .replace("webrtc://", "http://")
+        .replace("rtc://", "http://");
+
+    var vhost = a.hostname;
+    var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
+    var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
+
+    // parse the vhost in the params of app, that srs supports.
+    app = app.replace("...vhost...", "?vhost=");
+    if (app.indexOf("?") >= 0) {
+        var params = app.slice(app.indexOf("?"));
+        app = app.slice(0, app.indexOf("?"));
+
+        if (params.indexOf("vhost=") > 0) {
+            vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
+            if (vhost.indexOf("&") > 0) {
+                vhost = vhost.slice(0, vhost.indexOf("&"));
+            }
+        }
+    }
+
+    // when vhost equals to server, and server is ip,
+    // the vhost is __defaultVhost__
+    if (a.hostname === vhost) {
+        var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
+        if (re.test(a.hostname)) {
+            vhost = "__defaultVhost__";
+        }
+    }
+    
+    // parse the schema
+    var schema = "rtmp";
+    if (rtmp_url.indexOf("://") > 0) {
+        schema = rtmp_url.slice(0, rtmp_url.indexOf("://"));
+    }
+
+    var port = a.port;
+    if (!port) {
+        if (schema === 'http') {
+            port = 80;
+        } else if (schema === 'https') {
+            port = 443;
+        } else if (schema === 'rtmp') {
+            port = 1935;
+        }
+    }
+
+    var ret = {
+        url: rtmp_url,
+        schema: schema,
+        server: a.hostname, port: port,
+        vhost: vhost, app: app, stream: stream
+    };
+    __fill_query(a.search, ret);
+
+    // For webrtc API, we use 443 if page is https, or schema specified it.
+    if (!ret.port) {
+        if (schema === 'webrtc' || schema === 'rtc') {
+            if (ret.user_query.schema === 'https') {
+                ret.port = 443;
+            } else if (window.location.href.indexOf('https://') === 0) {
+                ret.port = 443;
+            } else {
+                // For WebRTC, SRS use 1985 as default API port.
+                ret.port = 1985;
+            }
+        }
+    }
+
+    return ret;
+}
+
+/**
+ * get the agent.
+ * @return an object specifies some browser.
+ *   for example, get_browser_agents().MSIE
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ */
+function get_browser_agents() {
+    var agent = navigator.userAgent;
+
+    /**
+     WindowsPC platform, Win7:
+     chrome 31.0.1650.63:
+     Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36
+     (KHTML, like Gecko) Chrome/31.0.1650.63 Safari/537.36
+     firefox 23.0.1:
+     Mozilla/5.0 (Windows NT 6.1; WOW64; rv:23.0) Gecko/20100101
+     Firefox/23.0
+     safari 5.1.7(7534.57.2):
+     Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/534.57.2
+     (KHTML, like Gecko) Version/5.1.7 Safari/534.57.2
+     opera 15.0.1147.153:
+     Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36
+     (KHTML, like Gecko) Chrome/28.0.1500.95 Safari/537.36
+     OPR/15.0.1147.153
+     360 6.2.1.272:
+     Mozilla/4.0 (compatible; MSIE 7.0; Windows NT 6.1; WOW64;
+     Trident/6.0; SLCC2; .NET CLR 2.0.50727; .NET CLR 3.5.30729;
+     .NET CLR 3.0.30729; Media Center PC 6.0; InfoPath.2; .NET4.0C;
+     .NET4.0E)
+     IE 10.0.9200.16750(update: 10.0.12):
+     Mozilla/4.0 (compatible; MSIE 7.0; Windows NT 6.1; WOW64;
+     Trident/6.0; SLCC2; .NET CLR 2.0.50727; .NET CLR 3.5.30729;
+     .NET CLR 3.0.30729; Media Center PC 6.0; InfoPath.2; .NET4.0C;
+     .NET4.0E)
+     */
+
+    return {
+        // platform
+        Android: agent.indexOf("Android") != -1,
+        Windows: agent.indexOf("Windows") != -1,
+        iPhone: agent.indexOf("iPhone") != -1,
+        // Windows Browsers
+        Chrome: agent.indexOf("Chrome") != -1,
+        Firefox: agent.indexOf("Firefox") != -1,
+        QQBrowser: agent.indexOf("QQBrowser") != -1,
+        MSIE: agent.indexOf("MSIE") != -1,
+        // Android Browsers
+        Opera: agent.indexOf("Presto") != -1,
+        MQQBrowser: agent.indexOf("MQQBrowser") != -1
+    };
+}
+
+/**
+ * format relative seconds to HH:MM:SS,
+ * for example, 210s formated to 00:03:30
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * @usage relative_seconds_to_HHMMSS(210)
+ */
+function relative_seconds_to_HHMMSS(seconds){
+    var date = new Date();
+    date.setTime(Number(seconds) * 1000);
+
+    var ret = padding(date.getUTCHours(), 2, '0')
+        + ":" + padding(date.getUTCMinutes(), 2, '0')
+        + ":" + padding(date.getUTCSeconds(), 2, '0');
+
+    return ret;
+}
+
+/**
+ * format absolute seconds to HH:MM:SS,
+ * for example, 1389146480s (2014-01-08 10:01:20 GMT+0800) formated to 10:01:20
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * @usage absolute_seconds_to_HHMMSS(new Date().getTime() / 1000)
+ */
+function absolute_seconds_to_HHMMSS(seconds){
+    var date = new Date();
+    date.setTime(Number(seconds) * 1000);
+
+    var ret = padding(date.getHours(), 2, '0')
+        + ":" + padding(date.getMinutes(), 2, '0')
+        + ":" + padding(date.getSeconds(), 2, '0');
+
+    return ret;
+}
+
+/**
+ * format absolute seconds to YYYY-mm-dd,
+ * for example, 1389146480s (2014-01-08 10:01:20 GMT+0800) formated to 2014-01-08
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * @usage absolute_seconds_to_YYYYmmdd(new Date().getTime() / 1000)
+ */
+function absolute_seconds_to_YYYYmmdd(seconds) {
+    var date = new Date();
+    date.setTime(Number(seconds) * 1000);
+
+    var ret = date.getFullYear()
+        + "-" + padding(date.getMonth() + 1, 2, '0')
+        + "-" + padding(date.getDate(), 2, '0');
+
+    return ret;
+}
+
+/**
+ * parse the date in str to Date object.
+ * @param str the date in str, format as "YYYY-mm-dd", for example, 2014-12-11
+ * @returns a date object.
+ * @usage YYYYmmdd_parse("2014-12-11")
+ */
+function YYYYmmdd_parse(str) {
+    var date = new Date();
+    date.setTime(Date.parse(str));
+    return date;
+}
+
+/**
+ * async refresh function call. to avoid multiple call.
+ * @remark AsyncRefresh is for jquery to refresh the speicified pfn in a page;
+ *      if angularjs, use AsyncRefresh2 to change pfn, cancel previous request for angularjs use singleton object.
+ * @param refresh_interval the default refresh interval ms.
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * the pfn can be implements as following:
+ var async_refresh = new AsyncRefresh(pfn, 3000);
+ function pfn() {
+            if (!async_refresh.refresh_is_enabled()) {
+                async_refresh.request(100);
+                return;
+            }
+            $.ajax({
+                type: 'GET', async: true, url: 'xxxxx',
+                complete: function(){
+                    if (!async_refresh.refresh_is_enabled()) {
+                        async_refresh.request(0);
+                    } else {
+                        async_refresh.request(async_refresh.refresh_interval);
+                    }
+                },
+                success: function(res){
+                    // if donot allow refresh, directly return.
+                    if (!async_refresh.refresh_is_enabled()) {
+                        return;
+                    }
+
+                    // render the res.
+                }
+            });
+        }
+ */
+function AsyncRefresh(pfn, refresh_interval) {
+    this.refresh_interval = refresh_interval;
+
+    this.__handler = null;
+    this.__pfn = pfn;
+
+    this.__enabled = true;
+}
+/**
+ * disable the refresher, the pfn must check the refresh state.
+ */
+AsyncRefresh.prototype.refresh_disable = function() {
+    this.__enabled = false;
+}
+AsyncRefresh.prototype.refresh_enable = function() {
+    this.__enabled = true;
+}
+AsyncRefresh.prototype.refresh_is_enabled = function() {
+    return this.__enabled;
+}
+/**
+ * start new async request
+ * @param timeout the timeout in ms.
+ *      user can use the refresh_interval of the AsyncRefresh object,
+ *      which initialized in constructor.
+ */
+AsyncRefresh.prototype.request = function(timeout) {
+    if (this.__handler) {
+        clearTimeout(this.__handler);
+    }
+
+    this.__handler = setTimeout(this.__pfn, timeout);
+}
+
+/**
+ * async refresh v2, support cancellable refresh, and change the refresh pfn.
+ * @remakr for angularjs. if user only need jquery, maybe AsyncRefresh is better.
+ * @see: http://blog.csdn.net/win_lin/article/details/17994347
+ * Usage:
+ bsmControllers.controller('CServers', ['$scope', 'MServer', function($scope, MServer){
+            async_refresh2.refresh_change(function(){
+                // 获取服务器列表
+                MServer.servers_load({}, function(data){
+                    $scope.servers = data.data.servers;
+                    async_refresh2.request();
+                });
+            }, 3000);
+
+            async_refresh2.request(0);
+        }]);
+ bsmControllers.controller('CStreams', ['$scope', 'MStream', function($scope, MStream){
+            async_refresh2.refresh_change(function(){
+                // 获取流列表
+                MStream.streams_load({}, function(data){
+                    $scope.streams = data.data.streams;
+                    async_refresh2.request();
+                });
+            }, 3000);
+
+            async_refresh2.request(0);
+        }]);
+ */
+function AsyncRefresh2() {
+    /**
+     * the function callback before call the pfn.
+     * the protype is function():bool, which return true to invoke, false to abort the call.
+     * null to ignore this callback.
+     *
+     * for example, user can abort the refresh by find the class popover:
+     *      async_refresh2.on_before_call_pfn = function() {
+     *          if ($(".popover").length > 0) {
+     *              async_refresh2.request();
+     *              return false;
+     *          }
+     *          return true;
+     *      };
+     */
+    this.on_before_call_pfn = null;
+
+    // use a anonymous function to call, and check the enabled when actually invoke.
+    this.__call = {
+        pfn: null,
+        timeout: 0,
+        __enabled: false,
+        __handler: null
+    };
+}
+// singleton
+var async_refresh2 = new AsyncRefresh2();
+/**
+ * initialize or refresh change. cancel previous request, setup new request.
+ * @param pfn a function():void to request after timeout. null to disable refresher.
+ * @param timeout the timeout in ms, to call pfn. null to disable refresher.
+ */
+AsyncRefresh2.prototype.initialize = function(pfn, timeout) {
+    this.refresh_change(pfn, timeout);
+}
+/**
+ * stop refresh, the refresh pfn is set to null.
+ */
+AsyncRefresh2.prototype.stop = function() {
+    this.__call.__enabled = false;
+}
+/**
+ * restart refresh, use previous config.
+ */
+AsyncRefresh2.prototype.restart = function() {
+    this.__call.__enabled = true;
+    this.request(0);
+}
+/**
+ * change refresh pfn, the old pfn will set to disabled.
+ */
+AsyncRefresh2.prototype.refresh_change = function(pfn, timeout) {
+    // cancel the previous call.
+    if (this.__call.__handler) {
+        clearTimeout(this.__handler);
+    }
+    this.__call.__enabled = false;
+
+    // setup new call.
+    this.__call = {
+        pfn: pfn,
+        timeout: timeout,
+        __enabled: true,
+        __handler: null
+    };
+}
+/**
+ * start new request, we never auto start the request,
+ * user must start new request when previous completed.
+ * @param timeout [optional] if not specified, use the timeout in initialize or refresh_change.
+ */
+AsyncRefresh2.prototype.request = function(timeout) {
+    var self = this;
+    var this_call = this.__call;
+
+    // clear previous timeout.
+    if (this_call.__handler) {
+        clearTimeout(this_call.__handler);
+    }
+
+    // override the timeout
+    if (timeout == undefined) {
+        timeout = this_call.timeout;
+    }
+
+    // if user disabled refresher.
+    if (this_call.pfn == null || timeout == null) {
+        return;
+    }
+
+    this_call.__handler = setTimeout(function(){
+        // cancelled by refresh_change, ignore.
+        if (!this_call.__enabled) {
+            return;
+        }
+
+        // callback if the handler installled.
+        if (self.on_before_call_pfn) {
+            if (!self.on_before_call_pfn()) {
+                return;
+            }
+        }
+
+        // do the actual call.
+        this_call.pfn();
+    }, timeout);
+}
+