【vue2】前端如何播放rtsp 视频流,拿到rtsp视频流地址如何处理,海康视频rtsp h264 如何播放

文章目录

    • 测试
    • 以vue2 为例
      • 新建 webrtcstreamer.js
      • 下载webrtc-streamer
      • video.vue
      • 页面中调用

        最近在写vue2 项目其中有个需求是实时播放摄像头的视频,摄像头是 海康的设备,搞了很长时间终于监控视频出来了,记录一下,放置下次遇到。文章有点长,略显啰嗦请耐心看完。

        测试

        测试?测试什么?测试rtsp视频流能不能播放。

        video mediaplay官网 即(VLC)

        下载、安装完VLC后,打开VLC 点击媒体 -> 打开网络串流

        将rtsp地址粘贴进去

        不能播放的话,rtsp视频流地址有问题。

        注意:视频可以播放也要查看视频的格式,如下

        右击视频选择工具->编解码器信息

        如果编解码是H264的,那么我的这种方法可以。如果是H265或者其他的话就要登录海康后台修改一下

        以vue2 为例

        新建 webrtcstreamer.js

        在public文件夹下新建webrtcstreamer.js文件,直接复制粘贴,无需修改

        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 { 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;
        }
        

        下载webrtc-streamer

        资源在最上面

        也可以去github上面下载:webrtc-streamer

        下载完后解压,打开,启动

        出现下面这个页面就是启动成功了,留意这里的端口号,就是我选出来的部分,一般都是默认8000,不排除其他情况

        检查一下也没用启动成功,http://127.0.0.1:8000/ 粘贴到浏览器地址栏回车查看,启动成功能看到电脑当前页面(这里的8000就是启动的端口号,启动的是多少就访问多少)

        video.vue

        新建video.js (位置自己决定,后面要引入的)

        video.js中要修改两个地方,第一个是引入webrtcstreamer.js路径,第二个地方是ip地址要要修改为自己的ip加上启动的端口号(即上面的8000),不知道电脑ip地址的看下面一行

        怎么查看自己的ip地址打开cmd 黑窗口(即dos窗口),输入ipconfig回车,在里面找到 IPv4 地址 就是了

        页面中调用

        在页面中引入video.vue,并注册。将rtsp视频地址传过去就好了,要显示几个视频就调用几次

        回到页面看,rtsp视频已经可以播放了