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Go语言开发浏览器视频流rtsp转webrtc播放

时间：2022-12-22 09:55:14|栏目：Golang|点击： 次

1. 前言

前面我们测试了rtsp转hls方式，发现延迟比较大，不太适合我们的使用需求。接下来我们试一下webrtc的方式看下使用情况。

综合考虑下来，我们最好能找到一个go作为后端，前端兼容性较好的前后端方案来处理webrtc，这样我们就可以结合我们之前的cgo+onvif+gSoap实现方案来获取rtsp流，并且可以根据已经实现的ptz、预置点等功能接口做更多的扩展。

2. rtsp转webRTC

如下是找到的一个比较合适的开源方案，前端使用了jQuery、bootstrap等，后端使用go+gin来实现并将rtsp流解析转换为webRTC协议提供http相关接口给到前端，通过config.json配置rtsp地址和stun地址：

点击下载

此外，还带有stun，可以自行配置stun地址，便于进行内网穿透。

初步测试几乎看不出来延迟，符合预期，使用的jQuery+bootstrap+go+gin做的web，也符合我们的项目使用情况。

3. 初步测试结果

结果如下：

4. 结合我们之前的onvif+gSoap+cgo的方案做修改

我们在此项目的基础上，结合我们之前研究的onvif+cgo+gSoap的方案，将onvif获取到的相关数据提供接口到web端，增加ptz、调焦、缩放等功能。

我们在http.go中添加新的post接口：HTTPAPIServerStreamPtz来进行ptz和HTTPAPIServerStreamPreset进行预置点相关操作。

以下是部分代码，没有做太多的优化，也仅仅实现了ptz、调焦和缩放，算是打通了通路，具体项目需要可以再做优化。

4.1 go后端修改

增加了新的接口，并将之前cgo+onvif+gSoap的内容结合了进来，项目整体没有做更多的优化，只是为了演示，提供一个思路：

http.go（增加了两个post接口ptz和preset，采用cgo方式处理）：

package main
/*
#cgo CFLAGS: -I ./ -I /usr/local/
#cgo LDFLAGS: -L ./build -lc_onvif_static -lpthread -ldl -lssl -lcrypto
#include "client.h"
#include "malloc.h"
*/
import "C"
import (
    "encoding/json"
    "fmt"
    "log"
    "net/http"
    "os"
    "sort"
    "strconv"
    "time"
    "unsafe"
    "github.com/deepch/vdk/av"
    webrtc "github.com/deepch/vdk/format/webrtcv3"
    "github.com/gin-gonic/gin"
)
type JCodec struct {
    Type string
}
func serveHTTP() {
    gin.SetMode(gin.ReleaseMode)
    router := gin.Default()
    router.Use(CORSMiddleware())
    if _, err := os.Stat("./web"); !os.IsNotExist(err) {
        router.LoadHTMLGlob("web/templates/*")
        router.GET("/", HTTPAPIServerIndex)
        router.GET("/stream/player/:uuid", HTTPAPIServerStreamPlayer)
    }
    router.POST("/stream/receiver/:uuid", HTTPAPIServerStreamWebRTC)
    //增加新的post接口
    router.POST("/stream/ptz/", HTTPAPIServerStreamPtz)
    router.POST("/stream/preset/", HTTPAPIServerStreamPreset)
    router.GET("/stream/codec/:uuid", HTTPAPIServerStreamCodec)
    router.POST("/stream", HTTPAPIServerStreamWebRTC2)
    router.StaticFS("/static", http.Dir("web/static"))
    err := router.Run(Config.Server.HTTPPort)
    if err != nil {
        log.Fatalln("Start HTTP Server error", err)
    }
}
//HTTPAPIServerIndex  index
func HTTPAPIServerIndex(c *gin.Context) {
    _, all := Config.list()
    if len(all) > 0 {
        c.Header("Cache-Control", "no-cache, max-age=0, must-revalidate, no-store")
        c.Header("Access-Control-Allow-Origin", "*")
        c.Redirect(http.StatusMovedPermanently, "stream/player/"+all[0])
    } else {
        c.HTML(http.StatusOK, "index.tmpl", gin.H{
            "port":    Config.Server.HTTPPort,
            "version": time.Now().String(),
        })
    }
}
//HTTPAPIServerStreamPlayer stream player
func HTTPAPIServerStreamPlayer(c *gin.Context) {
    _, all := Config.list()
    sort.Strings(all)
    c.HTML(http.StatusOK, "player.tmpl", gin.H{
        "port":     Config.Server.HTTPPort,
        "suuid":    c.Param("uuid"),
        "suuidMap": all,
        "version":  time.Now().String(),
    })
}
//HTTPAPIServerStreamCodec stream codec
func HTTPAPIServerStreamCodec(c *gin.Context) {
    if Config.ext(c.Param("uuid")) {
        Config.RunIFNotRun(c.Param("uuid"))
        codecs := Config.coGe(c.Param("uuid"))
        if codecs == nil {
            return
        }
        var tmpCodec []JCodec
        for _, codec := range codecs {
            if codec.Type() != av.H264 && codec.Type() != av.PCM_ALAW && codec.Type() != av.PCM_MULAW && codec.Type() != av.OPUS {
                log.Println("Codec Not Supported WebRTC ignore this track", codec.Type())
                continue
            }
            if codec.Type().IsVideo() {
                tmpCodec = append(tmpCodec, JCodec{Type: "video"})
            } else {
                tmpCodec = append(tmpCodec, JCodec{Type: "audio"})
            }
        }
        b, err := json.Marshal(tmpCodec)
        if err == nil {
			_, err = c.Writer.Write(b)
			if err != nil {
				log.Println("Write Codec Info error", err)
				return
			}
		}
	}
}
//HTTPAPIServerStreamWebRTC stream video over WebRTC
func HTTPAPIServerStreamWebRTC(c *gin.Context) {
	if !Config.ext(c.PostForm("suuid")) {
		log.Println("Stream Not Found")
		return
	}
	Config.RunIFNotRun(c.PostForm("suuid"))
	codecs := Config.coGe(c.PostForm("suuid"))
	if codecs == nil {
		log.Println("Stream Codec Not Found")
		return
	}
	var AudioOnly bool
	if len(codecs) == 1 && codecs[0].Type().IsAudio() {
		AudioOnly = true
	}
	muxerWebRTC := webrtc.NewMuxer(webrtc.Options{ICEServers: Config.GetICEServers(), ICEUsername: Config.GetICEUsername(), ICECredential: Config.GetICECredential(), PortMin: Config.GetWebRTCPortMin(), PortMax: Config.GetWebRTCPortMax()})
	answer, err := muxerWebRTC.WriteHeader(codecs, c.PostForm("data"))
	if err != nil {
		log.Println("WriteHeader", err)
		return
	}
	_, err = c.Writer.Write([]byte(answer))
	if err != nil {
		log.Println("Write", err)
		return
	}
	go func() {
		cid, ch := Config.clAd(c.PostForm("suuid"))
		defer Config.clDe(c.PostForm("suuid"), cid)
		defer muxerWebRTC.Close()
		var videoStart bool
		noVideo := time.NewTimer(10 * time.Second)
		for {
			select {
			case <-noVideo.C:
				log.Println("noVideo")
				return
			case pck := <-ch:
				if pck.IsKeyFrame || AudioOnly {
					noVideo.Reset(10 * time.Second)
					videoStart = true
				}
				if !videoStart && !AudioOnly {
					continue
				}
				err = muxerWebRTC.WritePacket(pck)
				if err != nil {
					log.Println("WritePacket", err)
					return
				}
			}
		}
	}()
}
func HTTPAPIServerStreamPtz(c *gin.Context) {
	action := c.PostForm("action")
	direction, err := strconv.Atoi(action)
	if err != nil {
		log.Println(err)
		return
	}
	var soap C.P_Soap
	soap = C.new_soap(soap)
	username := C.CString("admin")
	password := C.CString("admin")
	serviceAddr := C.CString("http://40.40.40.101:80/onvif/device_service")
	C.get_device_info(soap, username, password, serviceAddr)
	mediaAddr := [200]C.char{}
	C.get_capabilities(soap, username, password, serviceAddr, &mediaAddr[0])
	profileToken := [200]C.char{}
	C.get_profiles(soap, username, password, &profileToken[0], &mediaAddr[0])
	videoSourceToken := [200]C.char{}
	C.get_video_source(soap, username, password, &videoSourceToken[0], &mediaAddr[0])
	switch direction {
	case 0:
		break
	case 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11:
		C.ptz(soap, username, password, C.int(direction), C.float(0.5), &profileToken[0], &mediaAddr[0])
	case 12, 13, 14:
		C.focus(soap, username, password, C.int(direction), C.float(0.5), &videoSourceToken[0], &mediaAddr[0])
	default:
		fmt.Println("Unknown direction.")
	}
	C.del_soap(soap)
	C.free(unsafe.Pointer(username))
	C.free(unsafe.Pointer(password))
	C.free(unsafe.Pointer(serviceAddr))
	c.JSON(http.StatusOK, gin.H{"message":"success"})
}
func HTTPAPIServerStreamPreset(c *gin.Context) {
	var soap C.P_Soap
	soap = C.new_soap(soap)
	username := C.CString("admin")
	password := C.CString("admin")
	serviceAddr := C.CString("http://40.40.40.101:80/onvif/device_service")
	C.get_device_info(soap, username, password, serviceAddr)
	mediaAddr := [200]C.char{}
	C.get_capabilities(soap, username, password, serviceAddr, &mediaAddr[0])
	profileToken := [200]C.char{}
	C.get_profiles(soap, username, password, &profileToken[0], &mediaAddr[0])
	videoSourceToken := [200]C.char{}
	C.get_video_source(soap, username, password, &videoSourceToken[0], &mediaAddr[0])
	action := c.PostForm("action")
	presetAction, err := strconv.Atoi(action)
	if err != nil {
		log.Println(err)
		return
	}
	fmt.Println("请输入数字进行preset,1-4分别代表查询、设置、跳转、删除预置点；退出输入0:")
	switch presetAction {
	case 0:
		break
	case 1:
		C.preset(soap, username, password, C.int(presetAction), nil, nil, &profileToken[0], &mediaAddr[0])
	case 2:
		fmt.Println("请输入要设置的预置点token信息：")
		presentToken := ""
		_, _ = fmt.Scanln(&presentToken)
		fmt.Println("请输入要设置的预置点name信息长度不超过200：")
		presentName := ""
		_, _ = fmt.Scanln(&presentName)
		C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), C.CString(presentName), &profileToken[0], &mediaAddr[0])
	case 3:
		fmt.Println("请输入要跳转的预置点token信息：")
		presentToken := ""
		_, _ = fmt.Scanln(&presentToken)
		C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), nil, &profileToken[0], &mediaAddr[0])
	case 4:
		fmt.Println("请输入要删除的预置点token信息：")
		presentToken := ""
		_, _ = fmt.Scanln(&presentToken)
		C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), nil, &profileToken[0], &mediaAddr[0])
	default:
		fmt.Println("unknown present action.")
		break
	}
	C.del_soap(soap)
	C.free(unsafe.Pointer(username))
	C.free(unsafe.Pointer(password))
	C.free(unsafe.Pointer(serviceAddr))
	c.JSON(http.StatusOK, gin.H{"message":"success"})
}
func CORSMiddleware() gin.HandlerFunc {
	return func(c *gin.Context) {
		c.Header("Access-Control-Allow-Origin", "*")
		c.Header("Access-Control-Allow-Credentials", "true")
		c.Header("Access-Control-Allow-Headers", "Origin, X-Requested-With, Content-Type, Accept, Authorization, x-access-token")
		c.Header("Access-Control-Expose-Headers", "Content-Length, Access-Control-Allow-Origin, Access-Control-Allow-Headers, Cache-Control, Content-Language, Content-Type")
		c.Header("Access-Control-Allow-Methods", "POST, OPTIONS, GET, PUT")
		if c.Request.Method == "OPTIONS" {
			c.AbortWithStatus(http.StatusNoContent)
			return
		}
		c.Next()
	}
}
type Response struct {
	Tracks []string `json:"tracks"`
	Sdp64  string   `json:"sdp64"`
}
type ResponseError struct {
	Error string `json:"error"`
}
func HTTPAPIServerStreamWebRTC2(c *gin.Context) {
	url := c.PostForm("url")
	if _, ok := Config.Streams[url]; !ok {
		Config.Streams[url] = StreamST{
			URL:      url,
			OnDemand: true,
			Cl:       make(map[string]viewer),
		}
	}
	Config.RunIFNotRun(url)
	codecs := Config.coGe(url)
	if codecs == nil {
		log.Println("Stream Codec Not Found")
		c.JSON(500, ResponseError{Error: Config.LastError.Error()})
		return
	}
	muxerWebRTC := webrtc.NewMuxer(
		webrtc.Options{
			ICEServers: Config.GetICEServers(),
			PortMin:    Config.GetWebRTCPortMin(),
			PortMax:    Config.GetWebRTCPortMax(),
		},
	)
	sdp64 := c.PostForm("sdp64")
	answer, err := muxerWebRTC.WriteHeader(codecs, sdp64)
	if err != nil {
		log.Println("Muxer WriteHeader", err)
		c.JSON(500, ResponseError{Error: err.Error()})
		return
	}
	response := Response{
		Sdp64: answer,
	}
	for _, codec := range codecs {
		if codec.Type() != av.H264 &&
			codec.Type() != av.PCM_ALAW &&
			codec.Type() != av.PCM_MULAW &&
			codec.Type() != av.OPUS {
			log.Println("Codec Not Supported WebRTC ignore this track", codec.Type())
			continue
		}
		if codec.Type().IsVideo() {
			response.Tracks = append(response.Tracks, "video")
		} else {
			response.Tracks = append(response.Tracks, "audio")
		}
	}
	c.JSON(200, response)
	AudioOnly := len(codecs) == 1 && codecs[0].Type().IsAudio()
	go func() {
		cid, ch := Config.clAd(url)
		defer Config.clDe(url, cid)
		defer muxerWebRTC.Close()
		var videoStart bool
		noVideo := time.NewTimer(10 * time.Second)
		for {
			select {
			case <-noVideo.C:
				log.Println("noVideo")
				return
			case pck := <-ch:
				if pck.IsKeyFrame || AudioOnly {
					noVideo.Reset(10 * time.Second)
					videoStart = true
				}
				if !videoStart && !AudioOnly {
					continue
				}
				err = muxerWebRTC.WritePacket(pck)
				if err != nil {
					log.Println("WritePacket", err)
					return
				}
			}
		}
	}()
}

4.2 前端修改

对于goland我们首先将.tmpl文件通过右键标记为html格式，然后再修改时就会有前端语法支持和补全支持，便于修改，否则默认是识别为文本的，之后我们修改player.tmpl和app.js，在player.tmpl中添加一些ptz的按钮并通过js与前后端进行数据交互：

player.tmpl:

<html>
<meta http-equiv="Expires" content="0">
<meta http-equiv="Last-Modified" content="0">
<meta http-equiv="Cache-Control" content="no-cache, mustrevalidate">
<meta http-equiv="Pragma" content="no-cache">
<link rel="stylesheet" href="../../static/css/bootstrap.min.css" rel="external nofollow" >
<link rel="stylesheet" href="../../static/css/shanxing.css" rel="external nofollow" >
<script type="text/javascript" src="../../static/js/jquery-3.4.1.min.js"></script>
<script src="../../static/js/bootstrap.min.js"></script>
<script src="../../static/js/adapter-latest.js"></script>
<h2 align=center>
    Play Stream {{ .suuid }}<br>
</h2>
<div class="container">
    <div class="row">
        <div class="col-3">
            <div class="list-group">
                {{ range .suuidMap }}
                <a href="{{ . }}" rel="external nofollow"  id="{{ . }}" name="{{ . }}" class="list-group-item list-group-item-action">{{ . }}</a>
                {{ end }}
                </br>
                <div class="sector">
                    <div class="box s1" id="top" onclick="funTopClick()">
                    </div>
                    <div class="box s2" id="right" onclick="funRightClick()">
                    </div>
                    <div class="box s3" id="down" onclick="funDownClick()">
                    </div>
                    <div class="box s4" id="left" onclick="funLeftClick()">
                    </div>
                    <div class="center" id="stop" onclick="funStopClick()">
                    </div>
                </div>
                <div class="btn-group">
                    <button type="button" class="btn btn-default" onclick="funZoomClick(10)">缩放+</button>
                    <button type="button" class="btn btn-default" onclick="funZoomClick(11)">缩放-</button>
                </div>
                <div class="btn-group">
                    <button type="button" class="btn btn-default" onclick="funFocusClick(12)">调焦+</button>
                    <button type="button" class="btn btn-default" onclick="funFocusClick(13)">调焦-</button>
                    <button type="button" class="btn btn-default" onclick="funFocusClick(14)">停止调焦</button>
                </div>
            </div>
        </div>
        <div class="col">
            <input type="hidden" name="suuid" id="suuid" value="{{ .suuid }}">
            <input type="hidden" name="port" id="port" value="{{ .port }}">
            <input type="hidden" id="localSessionDescription" readonly="true">
            <input type="hidden" id="remoteSessionDescription">
            <div id="remoteVideos">
                <video style="width:600px" id="videoElem" autoplay muted controls></video>
            </div>
            <div id="div"></div>
        </div>
    </div>
</div>
<script type="text/javascript" src="../../static/js/app.js?ver={{ .version }}"></script>
</html>

app.js：

let stream = new MediaStream();
let suuid = $('#suuid').val();
let config = {
  iceServers: [{
    urls: ["stun:stun.l.google.com:19302"]
  }]
};
const pc = new RTCPeerConnection(config);
pc.onnegotiationneeded = handleNegotiationNeededEvent;
let log = msg => {
  document.getElementById('div').innerHTML += msg + '<br>'
}
pc.ontrack = function(event) {
  stream.addTrack(event.track);
  videoElem.srcObject = stream;
  log(event.streams.length + ' track is delivered')
}
pc.oniceconnectionstatechange = e => log(pc.iceConnectionState)
async function handleNegotiationNeededEvent() {
  let offer = await pc.createOffer();
  await pc.setLocalDescription(offer);
  getRemoteSdp();
}
$(document).ready(function() {
  $('#' + suuid).addClass('active');
  getCodecInfo();
});
function getCodecInfo() {
  $.get("../codec/" + suuid, function(data) {
    try {
      data = JSON.parse(data);
    } catch (e) {
      console.log(e);
    } finally {
      $.each(data,function(index,value){
        pc.addTransceiver(value.Type, {
          'direction': 'sendrecv'
        })
      })
    }
  });
}
let sendChannel = null;
function getRemoteSdp() {
  $.post("../receiver/"+ suuid, {
    suuid: suuid,
    data: btoa(pc.localDescription.sdp)
  }, function(data) {
    try {
      pc.setRemoteDescription(new RTCSessionDescription({
        type: 'answer',
        sdp: atob(data)
      }))
    } catch (e) {
      console.warn(e);
    }
  });
}
function ptz(direction) {
  $.post("../ptz/", direction, function(data, status){
    console.debug("Data: " + data + "nStatus: " + status);
  });
}
function funTopClick() {
  console.debug("top click");
  ptz("action=1")
}
function funDownClick() {
  console.debug("down click");
  ptz("action=2")
}
function funLeftClick() {
  console.debug("left click");
  ptz("action=3")
}
function funRightClick() {
  console.debug("right click");
  ptz("action=4")
}
function funStopClick() {
  console.debug("stop click");
  ptz("action=9")
}
function funZoomClick(direction) {
  console.debug("zoom click"+direction);
  ptz("action="+direction)
}
function funFocusClick(direction) {
  console.debug("focus click"+direction);
  ptz("action="+direction)
}

主要增加了一个扇形按钮和两组按钮组，然后将按钮的点击结合到app.js中进行处理，app.js中则发送post请求调用go后端接口。

4.3 项目结构和编译运行

项目结构如下，部分文件做了备份，实际可以不用：

$tree -a -I ".github|.idea|
build"
.
├── .gitignore
├── CMakeLists.txt
├── Dockerfile
├── LICENSE
├── README.md
├── build.cmd
├── client.c
├── client.h
├── config.go
├── config.json
├── config.json.bak
├── doc
│   ├── demo2.png
│   ├── demo3.png
│   └── demo4.png
├── go.mod
├── go.sum
├── http.go
├── main.go
├── main.go.bak
├── renovate.json
├── soap
│   ├── DeviceBinding.nsmap
│   ├── ImagingBinding.nsmap
│   ├── MediaBinding.nsmap
│   ├── PTZBinding.nsmap
│   ├── PullPointSubscriptionBinding.nsmap
│   ├── RemoteDiscoveryBinding.nsmap
│   ├── custom
│   │   ├── README.txt
│   │   ├── chrono_duration.cpp
│   │   ├── chrono_duration.h
│   │   ├── chrono_time_point.cpp
│   │   ├── chrono_time_point.h
│   │   ├── duration.c
│   │   ├── duration.h
│   │   ├── float128.c
│   │   ├── float128.h
│   │   ├── int128.c
│   │   ├── int128.h
│   │   ├── long_double.c
│   │   ├── long_double.h
│   │   ├── long_time.c
│   │   ├── long_time.h
│   │   ├── qbytearray_base64.cpp
│   │   ├── qbytearray_base64.h
│   │   ├── qbytearray_hex.cpp
│   │   ├── qbytearray_hex.h
│   │   ├── qdate.cpp
│   │   ├── qdate.h
│   │   ├── qdatetime.cpp
│   │   ├── qdatetime.h
│   │   ├── qstring.cpp
│   │   ├── qstring.h
│   │   ├── qtime.cpp
│   │   ├── qtime.h
│   │   ├── struct_timeval.c
│   │   ├── struct_timeval.h
│   │   ├── struct_tm.c
│   │   ├── struct_tm.h
│   │   ├── struct_tm_date.c
│   │   └── struct_tm_date.h
│   ├── dom.c
│   ├── dom.h
│   ├── duration.c
│   ├── duration.h
│   ├── mecevp.c
│   ├── mecevp.h
│   ├── onvif.h
│   ├── smdevp.c
│   ├── smdevp.h
│   ├── soapC.c
│   ├── soapClient.c
│   ├── soapH.h
│   ├── soapStub.h
│   ├── stdsoap2.h
│   ├── stdsoap2_ssl.c
│   ├── struct_timeval.c
│   ├── struct_timeval.h
│   ├── threads.c
│   ├── threads.h
│   ├── typemap.dat
│   ├── wsaapi.c
│   ├── wsaapi.h
│   ├── wsdd.nsmap
│   ├── wsseapi.c
│   └── wsseapi.h
├── stream.go
└── web
    ├── static
    │   ├── css
    │   │   ├── bootstrap-grid.css
    │   │   ├── bootstrap-grid.css.map
    │   │   ├── bootstrap-grid.min.css
    │   │   ├── bootstrap-grid.min.css.map
    │   │   ├── bootstrap-reboot.css
    │   │   ├── bootstrap-reboot.css.map
    │   │   ├── bootstrap-reboot.min.css
    │   │   ├── bootstrap-reboot.min.css.map
    │   │   ├── bootstrap.css
    │   │   ├── bootstrap.css.map
    │   │   ├── bootstrap.min.css
    │   │   ├── bootstrap.min.css.map
    │   │   └── shanxing.css
    │   └── js
    │       ├── adapter-latest.js
    │       ├── app.js
    │       ├── bootstrap.bundle.js
    │       ├── bootstrap.bundle.js.map
    │       ├── bootstrap.bundle.min.js
    │       ├── bootstrap.bundle.min.js.map
    │       ├── bootstrap.js
    │       ├── bootstrap.js.map
    │       ├── bootstrap.min.js
    │       ├── bootstrap.min.js.map
    │       └── jquery-3.4.1.min.js
    └── templates
        ├── index.tmpl
        └── player.tmpl
8 directories, 111 files

关于cgo和onvif、gSoap部分这里就不多说了，不清楚的可以看前面的总结，gin、bootstramp、jQuery这些也需要一定的前后端概念学习和储备，在其它的分类总结中也零星分布了，不清楚的可以看一下，这里就不再多说了。

编译运行：

GOOS=linux GOARCH=amd64 CGO_ENABLE=1 GO111MODULE=on go run *.go

记得修改一下go.mod中对go版本的依赖，按照cgo的问题，目前至少需要1.18及以上，否则运行ptz可能出现分割违例问题，到我总结这里1.18已经发了正式版本了。

module github.com/deepch/RTSPtoWebRTC
go 1.18
require (
	github.com/deepch/vdk v0.0.0-20220309163430-c6529706436c
	github.com/gin-gonic/gin v1.7.7
)

4.4 结果展示

界面效果：

动态调试ptz：

动态调试缩放：

动态调试调焦：

5. 最后

webRTC使用起来几乎感觉不到延迟，但是受制于stun的udp打洞的稳定性，可能会出现卡顿掉线等情况，所以还牵扯到p2p的问题，需要注意这一点，当然，这是远程推流都绕不开的一点，也不算是独有的问题。

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本文标题：Go语言开发浏览器视频流rtsp转webrtc播放

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