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
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; }
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)
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
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