vue2使用rtsp视频流接入海康威视摄像头(纯前端)

🕗 发布于 2024-12-11 06:17 前端

一.获取海康威视rtsp视频流

海康威视官方的RTSP最新取流格式如下:

rtsp://用户名:密码@IP:554/Streaming/Channels/101

用户名和密码

IP就是登陆摄像头时候的IP(笔者这里IP是192.168.1.210)

所以笔者的rtsp流地址就是rtsp://用户名:密码@192.168.1.210:554/Streaming/Channels/101

二. 测试rtsp流是否可以播放

1.实现RTSP协议推流需要做的配置

1.1关闭萤石云的接入

1.2调整视频编码为H.264

2.安装VLC播放器

在此下载 video mediaplay官网即（VLC）

安装完成之后打开VLC播放器

在VLC播放器中打开网络串流输入rtsp地址

成功的话我们可以看到我们所显示的摄像头

如果RTSP流地址正确且取流成功，VLC的界面会显示监控画面。否则会报错，报错信息写在了日志里，在[工具]>[消息]里可以看到

三.在vue2中引用rtsp视频流形式的海康摄像头

1.新建webrtcstreamer.js文件

在public文件夹下新建webrtcstreamer.js 代码贴在下方,复制粘贴即可

var WebRtcStreamer = (function() {

/** 
 * Interface with WebRTC-streamer API
 * @constructor
 * @param {string} videoElement - id of the video element tag
 * @param {string} srvurl -  url of webrtc-streamer (default is current location)
*/
var WebRtcStreamer = function WebRtcStreamer (videoElement, srvurl) {
if (typeof videoElement === "string") {
this.videoElement = document.getElementById(videoElement);
} else {
this.videoElement = videoElement;
}
this.srvurl           = srvurl || location.protocol+"//"+window.location.hostname+":"+window.location.port;
this.pc               = null;    

this.mediaConstraints = { offerToReceiveAudio: true, offerToReceiveVideo: true };

this.iceServers = null;
this.earlyCandidates = [];
}

WebRtcStreamer.prototype._handleHttpErrors = function (response) {
    if (!response.ok) {
        throw Error(response.statusText);
    }
    return response;
}

/** 
 * Connect a WebRTC Stream to videoElement 
 * @param {string} videourl - id of WebRTC video stream
 * @param {string} audiourl - id of WebRTC audio stream
 * @param {string} options -  options of WebRTC call
 * @param {string} stream  -  local stream to send
*/
WebRtcStreamer.prototype.connect = function(videourl, audiourl, options, localstream) {
this.disconnect();

// getIceServers is not already received
if (!this.iceServers) {
console.log("Get IceServers");

fetch(this.srvurl + "/api/getIceServers")
.then(this._handleHttpErrors)
.then( (response) => (response.json()) )
.then( (response) =>  this.onReceiveGetIceServers(response, videourl, audiourl, options, localstream))
.catch( (error) => this.onError("getIceServers " + error ))

} else {
this.onReceiveGetIceServers(this.iceServers, videourl, audiourl, options, localstream);
}
}

/** 
 * Disconnect a WebRTC Stream and clear videoElement source
*/
WebRtcStreamer.prototype.disconnect = function() {
if (this.videoElement?.srcObject) {
this.videoElement.srcObject.getTracks().forEach(track => {
track.stop()
this.videoElement.srcObject.removeTrack(track);
});
}
if (this.pc) {
fetch(this.srvurl + "/api/hangup?peerid=" + this.pc.peerid)
.then(this._handleHttpErrors)
.catch( (error) => this.onError("hangup " + error ))


try {
this.pc.close();
}
catch (e) {
console.log ("Failure close peer connection:" + e);
}
this.pc = null;
}
}    

/*
* GetIceServers callback
*/
WebRtcStreamer.prototype.onReceiveGetIceServers = function(iceServers, videourl, audiourl, options, stream) {
this.iceServers       = iceServers;
this.pcConfig         = iceServers || {"iceServers": [] };
try {            
this.createPeerConnection();

var callurl = this.srvurl + "/api/call?peerid=" + this.pc.peerid + "&url=" + encodeURIComponent(videourl);
if (audiourl) {
callurl += "&audiourl="+encodeURIComponent(audiourl);
}
if (options) {
callurl += "&options="+encodeURIComponent(options);
}

if (stream) {
this.pc.addStream(stream);
}

                // clear early candidates
this.earlyCandidates.length = 0;

// create Offer
this.pc.createOffer(this.mediaConstraints).then((sessionDescription) => {
console.log("Create offer:" + JSON.stringify(sessionDescription));

this.pc.setLocalDescription(sessionDescription)
.then(() => {
fetch(callurl, { method: "POST", body: JSON.stringify(sessionDescription) })
.then(this._handleHttpErrors)
.then( (response) => (response.json()) )
.catch( (error) => this.onError("call " + error ))
.then( (response) =>  this.onReceiveCall(response) )
.catch( (error) => this.onError("call " + error ))

}, (error) => {
console.log ("setLocalDescription error:" + JSON.stringify(error)); 
});

}, (error) => { 
alert("Create offer error:" + JSON.stringify(error));
});

} catch (e) {
this.disconnect();
alert("connect error: " + e);
}    
}


WebRtcStreamer.prototype.getIceCandidate = function() {
fetch(this.srvurl + "/api/getIceCandidate?peerid=" + this.pc.peerid)
.then(this._handleHttpErrors)
.then( (response) => (response.json()) )
.then( (response) =>  this.onReceiveCandidate(response))
.catch( (error) => this.onError("getIceCandidate " + error ))
}

/*
* create RTCPeerConnection 
*/
WebRtcStreamer.prototype.createPeerConnection = function() {
console.log("createPeerConnection  config: " + JSON.stringify(this.pcConfig));
this.pc = new RTCPeerConnection(this.pcConfig);
var pc = this.pc;
pc.peerid = Math.random();

pc.onicecandidate = (evt) => this.onIceCandidate(evt);
pc.onaddstream    = (evt) => this.onAddStream(evt);
pc.oniceconnectionstatechange = (evt) => {  
console.log("oniceconnectionstatechange  state: " + pc.iceConnectionState);
if (this.videoElement) {
if (pc.iceConnectionState === "connected") {
this.videoElement.style.opacity = "1.0";
}
else if (pc.iceConnectionState === "disconnected") {
this.videoElement.style.opacity = "0.25";
}
else if ( (pc.iceConnectionState === "failed") || (pc.iceConnectionState === "closed") )  {
this.videoElement.style.opacity = "0.5";
} else if (pc.iceConnectionState === "new") {
this.getIceCandidate();
}
}
}
pc.ondatachannel = function(evt) {  
console.log("remote datachannel created:"+JSON.stringify(evt));

evt.channel.onopen = function () {
console.log("remote datachannel open");
this.send("remote channel openned");
}
evt.channel.onmessage = function (event) {
console.log("remote datachannel recv:"+JSON.stringify(event.data));
}
}
pc.onicegatheringstatechange = function() {
if (pc.iceGatheringState === "complete") {
const recvs = pc.getReceivers();

recvs.forEach((recv) => {
  if (recv.track && recv.track.kind === "video") {
console.log("codecs:" + JSON.stringify(recv.getParameters().codecs))
  }
});
  }
}

try {
var dataChannel = pc.createDataChannel("ClientDataChannel");
dataChannel.onopen = function() {
console.log("local datachannel open");
this.send("local channel openned");
}
dataChannel.onmessage = function(evt) {
console.log("local datachannel recv:"+JSON.stringify(evt.data));
}
} catch (e) {
console.log("Cannor create datachannel error: " + e);
}

console.log("Created RTCPeerConnnection with config: " + JSON.stringify(this.pcConfig) );
return pc;
}


/*
* RTCPeerConnection IceCandidate callback
*/
WebRtcStreamer.prototype.onIceCandidate = function (event) {
if (event.candidate) {
if (this.pc.currentRemoteDescription)  {
this.addIceCandidate(this.pc.peerid, event.candidate);
} else {
this.earlyCandidates.push(event.candidate);
}
} 
else {
console.log("End of candidates.");
}
}


WebRtcStreamer.prototype.addIceCandidate = function(peerid, candidate) {
fetch(this.srvurl + "/api/addIceCandidate?peerid="+peerid, { method: "POST", body: JSON.stringify(candidate) })
.then(this._handleHttpErrors)
.then( (response) => (response.json()) )
.then( (response) =>  {console.log("addIceCandidate ok:" + response)})
.catch( (error) => this.onError("addIceCandidate " + error ))
}

/*
* RTCPeerConnection AddTrack callback
*/
WebRtcStreamer.prototype.onAddStream = function(event) {
console.log("Remote track added:" +  JSON.stringify(event));

this.videoElement.srcObject = event.stream;
var promise = this.videoElement.play();
if (promise !== undefined) {
  promise.catch((error) => {
console.warn("error:"+error);
this.videoElement.setAttribute("controls", true);
  });
}
}

/*
* AJAX /call callback
*/
WebRtcStreamer.prototype.onReceiveCall = function(dataJson) {

console.log("offer: " + JSON.stringify(dataJson));
var descr = new RTCSessionDescription(dataJson);
this.pc.setRemoteDescription(descr).then(() =>  { 
console.log ("setRemoteDescription ok");
while (this.earlyCandidates.length) {
var candidate = this.earlyCandidates.shift();
this.addIceCandidate(this.pc.peerid, candidate);
}

this.getIceCandidate()
}
, (error) => { 
console.log ("setRemoteDescription error:" + JSON.stringify(error)); 
});
}

/*
* AJAX /getIceCandidate callback
*/
WebRtcStreamer.prototype.onReceiveCandidate = function(dataJson) {
console.log("candidate: " + JSON.stringify(dataJson));
if (dataJson) {
for (var i=0; i<dataJson.length; i++) {
var candidate = new RTCIceCandidate(dataJson[i]);

console.log("Adding ICE candidate :" + JSON.stringify(candidate) );
this.pc.addIceCandidate(candidate).then( () =>      { console.log ("addIceCandidate OK"); }
, (error) => { console.log ("addIceCandidate error:" + JSON.stringify(error)); } );
}
this.pc.addIceCandidate();
}
}


/*
* AJAX callback for Error
*/
WebRtcStreamer.prototype.onError = function(status) {
console.log("onError:" + status);
}

return WebRtcStreamer;
})();

if (typeof window !== 'undefined' && typeof window.document !== 'undefined') {
window.WebRtcStreamer = WebRtcStreamer;
}
if (typeof module !== 'undefined' && typeof module.exports !== 'undefined') {
module.exports = WebRtcStreamer;
}

2.下载webrtc-streamer

资源在最上面
也可以去github上面下载：webrtc-streamer

下载完解压,打开文件夹,启动webrtc-streamer.exe

打开完会出现cmd一样的黑框框如下

如果没有启动成功可以在浏览器中输入http://127.0.0.1:8000/查看本地端口8000是否被其他应用程序占用，如果没有被占用打开窗口应该如下图所示(是可以看见自己的页面的)

3.封装组件video.vue(名字随意)

代码如下(但是有需要注意的地方，请看下方)

<template>
  <div id="video-contianer">
    <video
      class="video"
      ref="video"
      preload="auto"
      autoplay="autoplay"
      muted
      width="600"
      height="400"
    />
    <div
      class="mask"
      @click="handleClickVideo"
      :class="{ 'active-video-border': selectStatus }"
    ></div>
  </div>
</template>

<script>
import WebRtcStreamer from '../../public/hk/webrtcstreamer'

export default {
  name: 'videoCom',
  props: {
    rtsp: {
      type: String,
      required: true,
    },
    isOn: {
      type: Boolean,
      default: false,
    },
    spareId: {
      type: Number,
    },
    selectStatus: {
      type: Boolean,
      default: false,
    },
  },
  data() {
    return {
      socket: null,
      result: null, // 返回值
      pic: null,
      webRtcServer: null,
      clickCount: 0, // 用来计数点击次数
    }
  },
  watch: {
    rtsp() {
      // do something
      console.log(this.rtsp)
      this.webRtcServer.disconnect()
      this.initVideo()
    },
  },
  destroyed() {
    this.webRtcServer.disconnect()
  },
  beforeCreate() {
    window.onbeforeunload = () => {
      this.webRtcServer.disconnect()
    }
  },
  created() {},
  mounted() {
    this.initVideo()
  },
  methods: {
    initVideo() {
      try {
        //连接后端的IP地址和端口
        this.webRtcServer = new WebRtcStreamer(
          this.$refs.video,
          `http://192.168.1.102:8000`
        )
        //向后端发送rtsp地址
        this.webRtcServer.connect(this.rtsp)
      } catch (error) {
        console.log(error)
      }
    },
    /* 处理双击 单机 */
    dbClick() {
      this.clickCount++
      if (this.clickCount === 2) {
        this.btnFull() // 双击全屏
        this.clickCount = 0
      }
      setTimeout(() => {
        if (this.clickCount === 1) {
          this.clickCount = 0
        }
      }, 250)
    },
    /* 视频全屏 */
    btnFull() {
      const elVideo = this.$refs.video
      if (elVideo.webkitRequestFullScreen) {
        elVideo.webkitRequestFullScreen()
      } else if (elVideo.mozRequestFullScreen) {
        elVideo.mozRequestFullScreen()
      } else if (elVideo.requestFullscreen) {
        elVideo.requestFullscreen()
      }
    },
    /* 
    ison用来判断是否需要更换视频流
    dbclick函数用来双击放大全屏方法
    */
    handleClickVideo() {
      if (this.isOn) {
        this.$emit('selectVideo', this.spareId)
        this.dbClick()
      } else {
        this.btnFull()
      }
    },
  },
}
</script>

<style scoped lang="scss">
.active-video-border {
  border: 2px salmon solid;
}
#video-contianer {
  position: relative;
  // width: 100%;
  // height: 100%;
  .video {
    // width: 100%;
    // height: 100%;
    // object-fit: cover;
  }
  .mask {
    position: absolute;
    top: 0;
    left: 0;
    width: 100%;
    height: 100%;
    cursor: pointer;
  }
}
</style>

这里要注意两个地方

第一个是

第二个是

不会查看本机端口的看这里(首先使用 Win + R打开运行输入cmd)

4.使用video封装组件播放rtsp视频流

首先我们在要使用video封装组件的地方引入并且注册video组件

之后在页面中使用video组件并且定义了两个变量将rtsp流传给封装的video组件

效果图如下

5.使用此种方法播放的时候会默认带声音播放,如何取消(看这里)

之后声明一个方法

然后在created里面调用就静音了

四.其他功能

1.截图功能

rtsp流引入海康威视摄像头——截图功能-CSDN博客

到此为止海康摄像头引入vue的方法就完美完结了

如果同学们有什么好的意见或者有什么问题可以私信我

最后祝大家事业蒸蒸日上,心想事成！

原文地址：https://blog.csdn.net/m0_74825614/article/details/144386234

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