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vue视频直接播放rtsp流；vue视频延迟问题解决；webRTC占cpu太大卡死问题解决；解决webRTC播放卡花屏问题:_webrtcstreamer.prototype.onerror设置

作者：IT小白 | 2024-06-19 05:49:38

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webrtcstreamer.prototype.onerror设置

播放多个视频

  1.  <div class="video-box">
  2.       <div class="video">
  3.         <iframe style="width:100%;height:100%;" name="ddddd" id="iframes" scrolling="auto" :src="videoLeftUrl"></iframe>
  4.       </div>
  5.       <div class="video">
  6.         <iframe style="width:100%;height:100%;" name="ddddd" id="iframes" scrolling="auto" :src="videoRightUrl"></iframe>
  7.       </div>
  8.       <div class="video">
  9.         <iframe style="width:100%;height:100%;" name="ddddd" id="iframes" scrolling="auto" :src="videoRtspUrl"></iframe>
  10.       </div>
  11.     </div>

js部分其中的item就是rtsp视频流

  1.     getShareVideoLeftUrl(item) {
  2.       this.videoLeftUrl = `/static/test.html?data=${item}`
  3.     },
  4.     getShareVideoRightUrl(item) {
  5.       this.videoRightUrl = `/static/test.html?data=${item}`
  6.     },
  7.     getShareVideoRtspUrl(item) {
  8.       this.videoRtspUrl = `/static/test.html?data=${item}`
  9.     },

public/static/test.html内容

  1. <html>
  2. 	<head>
  3. 		<script src="js/webrtcstreamer.js"></script>
  4. 		<script>
  5. 			// 接受从vue组件中传过来的参数
  6. 			let url = location.search; //这一条语句获取了包括问号开始到参数的最后，不包括前面的路径
  7. 			let params = url.substr(1); //去掉问号
  8. 			let pa = params.split("&");
  9. 			let s = new Object();
  10.              //  设置后端服务地址
  11.             let VIDEOURL = "http://172.18.127.7:8000" //服务视频webrtc
  12.  
  13. 			for (let i = 0; i < pa.length; i++) {
  14. 				s[pa[i].split("=")[0]] = unescape(pa[i].split("=")[1]);
  15. 			}
  16. 			console.log(s.data)
  17. 			window.onload = function() {
  18. 				webRtcServer = new WebRtcStreamer("video", VIDEOURL);
  19. 				webRtcServer.connect(s.data);
  20. 			}
  21. 			window.onbeforeunload = function() {
  22. 				webRtcServer.disconnect();
  23. 			}
  24. 		</script>
  25. 	</head>
  26. 	<body>
  27. 		<h1 value="da3"></h1>
  28. 		<video id="video" style="width: 100%;height: 100%;" controls autoplay muted />
  29. 	</body>
  30. </html>
  31.

其中public/static/js/webrtcstreamer.js文件内容如下

  1. var WebRtcStreamer = (function() {
  2.  
  3. /** 
  4.  * Interface with WebRTC-streamer API
  5.  * @constructor
  6.  * @param {string} videoElement - id of the video element tag
  7.  * @param {string} srvurl -  url of webrtc-streamer (default is current location)
  8. */
  9. var WebRtcStreamer = function WebRtcStreamer (videoElement, srvurl) {
  10. 	if (typeof videoElement === "string") {
  11. 		this.videoElement = document.getElementById(videoElement);
  12. 	} else {
  13. 		this.videoElement = videoElement;
  14. 	}
  15. 	this.srvurl           = srvurl || location.protocol+"//"+window.location.hostname+":"+window.location.port;
  16. 	this.pc               = null;    
  17.  
  18. 	this.pcOptions        = { "optional": [{"DtlsSrtpKeyAgreement": true} ] };
  19.  
  20. 	this.mediaConstraints = { offerToReceiveAudio: true, offerToReceiveVideo: true };
  21.  
  22. 	this.iceServers = null;
  23. 	this.earlyCandidates = [];
  24. }
  25.  
  26. WebRtcStreamer.prototype._handleHttpErrors = function (response) {
  27.     if (!response.ok) {
  28.         throw Error(response.statusText);
  29.     }
  30.     return response;
  31. }
  32.  
  33. /** 
  34.  * Connect a WebRTC Stream to videoElement 
  35.  * @param {string} videourl - id of WebRTC video stream
  36.  * @param {string} audiourl - id of WebRTC audio stream
  37.  * @param {string} options -  options of WebRTC call
  38.  * @param {string} stream  -  local stream to send
  39. */
  40. WebRtcStreamer.prototype.connect = function(videourl, audiourl, options, localstream) {
  41. 	this.disconnect();
  42. 	
  43. 	// getIceServers is not already received
  44. 	if (!this.iceServers) {
  45. 		console.log("Get IceServers");
  46. 		
  47. 		fetch(this.srvurl + "/api/getIceServers")
  48. 			.then(this._handleHttpErrors)
  49. 			.then( (response) => (response.json()) )
  50. 			.then( (response) =>  this.onReceiveGetIceServers.call(this,response, videourl, audiourl, options, localstream))
  51. 			.catch( (error) => this.onError("getIceServers " + error ))
  52. 				
  53. 	} else {
  54. 		this.onReceiveGetIceServers(this.iceServers, videourl, audiourl, options, localstream);
  55. 	}
  56. }
  57.  
  58. /** 
  59.  * Disconnect a WebRTC Stream and clear videoElement source
  60. */
  61. WebRtcStreamer.prototype.disconnect = function() {		
  62. 	if (this.videoElement) {
  63. 		this.videoElement.src = "";
  64. 	}
  65. 	if (this.pc) {
  66. 		fetch(this.srvurl + "/api/hangup?peerid="+this.pc.peerid)
  67. 			.then(this._handleHttpErrors)
  68. 			.catch( (error) => this.onError("hangup " + error ))
  69.  
  70. 		
  71. 		try {
  72. 			this.pc.close();
  73. 		}
  74. 		catch (e) {
  75. 			console.log ("Failure close peer connection:" + e);
  76. 		}
  77. 		this.pc = null;
  78. 	}
  79. }    
  80.  
  81. /*
  82. * GetIceServers callback
  83. */
  84. WebRtcStreamer.prototype.onReceiveGetIceServers = function(iceServers, videourl, audiourl, options, stream) {
  85. 	this.iceServers       = iceServers;
  86. 	this.pcConfig         = iceServers || {"iceServers": [] };
  87. 	try {            
  88. 		this.createPeerConnection();
  89.  
  90. 		var callurl = this.srvurl + "/api/call?peerid="+ this.pc.peerid+"&url="+encodeURIComponent(videourl);
  91. 		if (audiourl) {
  92. 			callurl += "&audiourl="+encodeURIComponent(audiourl);
  93. 		}
  94. 		if (options) {
  95. 			callurl += "&options="+encodeURIComponent(options);
  96. 		}
  97. 		
  98. 		if (stream) {
  99. 			this.pc.addStream(stream);
  100. 		}
  101.  
  102.                 // clear early candidates
  103. 		this.earlyCandidates.length = 0;
  104. 		
  105. 		// create Offer
  106. 		var bind = this;
  107. 		this.pc.createOffer(this.mediaConstraints).then(function(sessionDescription) {
  108. 			console.log("Create offer:" + JSON.stringify(sessionDescription));
  109. 			
  110. 			bind.pc.setLocalDescription(sessionDescription
  111. 				, function() {
  112. 					fetch(callurl, { method: "POST", body: JSON.stringify(sessionDescription) })
  113. 						.then(bind._handleHttpErrors)
  114. 						.then( (response) => (response.json()) )
  115. 						.catch( (error) => bind.onError("call " + error ))
  116. 						.then( (response) =>  bind.onReceiveCall.call(bind,response) )
  117. 						.catch( (error) => bind.onError("call " + error ))
  118. 				
  119. 				}
  120. 				, function(error) {
  121. 					console.log ("setLocalDescription error:" + JSON.stringify(error)); 
  122. 				} );
  123. 			
  124. 		}, function(error) { 
  125. 			alert("Create offer error:" + JSON.stringify(error));
  126. 		});
  127.  
  128. 	} catch (e) {
  129. 		this.disconnect();
  130. 		alert("connect error: " + e);
  131. 	}	    
  132. }
  133.  
  134.  
  135. WebRtcStreamer.prototype.getIceCandidate = function() {
  136. 	fetch(this.srvurl + "/api/getIceCandidate?peerid=" + this.pc.peerid)
  137. 		.then(this._handleHttpErrors)
  138. 		.then( (response) => (response.json()) )
  139. 		.then( (response) =>  this.onReceiveCandidate.call(this, response))
  140. 		.catch( (error) => bind.onError("getIceCandidate " + error ))
  141. }
  142. 					
  143. /*
  144. * create RTCPeerConnection 
  145. */
  146. WebRtcStreamer.prototype.createPeerConnection = function() {
  147. 	console.log("createPeerConnection  config: " + JSON.stringify(this.pcConfig) + " option:"+  JSON.stringify(this.pcOptions));
  148. 	this.pc = new RTCPeerConnection(this.pcConfig, this.pcOptions);
  149. 	var pc = this.pc;
  150. 	pc.peerid = Math.random();		
  151. 	
  152. 	var bind = this;
  153. 	pc.onicecandidate = function(evt) { bind.onIceCandidate.call(bind, evt); };
  154. 	pc.onaddstream    = function(evt) { bind.onAddStream.call(bind,evt); };
  155. 	pc.oniceconnectionstatechange = function(evt) {  
  156. 		console.log("oniceconnectionstatechange  state: " + pc.iceConnectionState);
  157. 		if (bind.videoElement) {
  158. 			if (pc.iceConnectionState === "connected") {
  159. 				bind.videoElement.style.opacity = "1.0";
  160. 			}			
  161. 			else if (pc.iceConnectionState === "disconnected") {
  162. 				bind.videoElement.style.opacity = "0.25";
  163. 			}			
  164. 			else if ( (pc.iceConnectionState === "failed") || (pc.iceConnectionState === "closed") )  {
  165. 				bind.videoElement.style.opacity = "0.5";
  166. 			} else if (pc.iceConnectionState === "new") {
  167. 				bind.getIceCandidate.call(bind)
  168. 			}
  169. 		}
  170. 	}
  171. 	pc.ondatachannel = function(evt) {  
  172. 		console.log("remote datachannel created:"+JSON.stringify(evt));
  173. 		
  174. 		evt.channel.onopen = function () {
  175. 			console.log("remote datachannel open");
  176. 			this.send("remote channel openned");
  177. 		}
  178. 		evt.channel.onmessage = function (event) {
  179. 			console.log("remote datachannel recv:"+JSON.stringify(event.data));
  180. 		}
  181. 	}
  182. 	pc.onicegatheringstatechange = function() {
  183. 		if (pc.iceGatheringState === "complete") {
  184. 			const recvs = pc.getReceivers();
  185. 		
  186. 			recvs.forEach((recv) => {
  187. 			  if (recv.track && recv.track.kind === "video") {
  188. 				console.log("codecs:" + JSON.stringify(recv.getParameters().codecs))
  189. 			  }
  190. 			});
  191. 		  }
  192. 	}
  193.  
  194. 	try {
  195. 		var dataChannel = pc.createDataChannel("ClientDataChannel");
  196. 		dataChannel.onopen = function() {
  197. 			console.log("local datachannel open");
  198. 			this.send("local channel openned");
  199. 		}
  200. 		dataChannel.onmessage = function(evt) {
  201. 			console.log("local datachannel recv:"+JSON.stringify(evt.data));
  202. 		}
  203. 	} catch (e) {
  204. 		console.log("Cannor create datachannel error: " + e);
  205. 	}	
  206. 	
  207. 	console.log("Created RTCPeerConnnection with config: " + JSON.stringify(this.pcConfig) + "option:"+  JSON.stringify(this.pcOptions) );
  208. 	return pc;
  209. }
  210.  
  211.  
  212. /*
  213. * RTCPeerConnection IceCandidate callback
  214. */
  215. WebRtcStreamer.prototype.onIceCandidate = function (event) {
  216. 	if (event.candidate) {
  217. 		if (this.pc.currentRemoteDescription)  {
  218. 			this.addIceCandidate(this.pc.peerid, event.candidate);					
  219. 		} else {
  220. 			this.earlyCandidates.push(event.candidate);
  221. 		}
  222. 	} 
  223. 	else {
  224. 		console.log("End of candidates.");
  225. 	}
  226. }
  227.  
  228.  
  229. WebRtcStreamer.prototype.addIceCandidate = function(peerid, candidate) {
  230. 	fetch(this.srvurl + "/api/addIceCandidate?peerid="+peerid, { method: "POST", body: JSON.stringify(candidate) })
  231. 		.then(this._handleHttpErrors)
  232. 		.then( (response) => (response.json()) )
  233. 		.then( (response) =>  {console.log("addIceCandidate ok:" + response)})
  234. 		.catch( (error) => this.onError("addIceCandidate " + error ))
  235. }
  236. 				
  237. /*
  238. * RTCPeerConnection AddTrack callback
  239. */
  240. WebRtcStreamer.prototype.onAddStream = function(event) {
  241. 	console.log("Remote track added:" +  JSON.stringify(event));
  242. 	
  243. 	this.videoElement.srcObject = event.stream;
  244. 	var promise = this.videoElement.play();
  245. 	if (promise !== undefined) {
  246. 	  var bind = this;
  247. 	  promise.catch(function(error) {
  248. 		console.warn("error:"+error);
  249. 		bind.videoElement.setAttribute("controls", true);
  250. 	  });
  251. 	}
  252. }
  253. 		
  254. /*
  255. * AJAX /call callback
  256. */
  257. WebRtcStreamer.prototype.onReceiveCall = function(dataJson) {
  258. 	var bind = this;
  259. 	console.log("offer: " + JSON.stringify(dataJson));
  260. 	var descr = new RTCSessionDescription(dataJson);
  261. 	this.pc.setRemoteDescription(descr
  262. 		, function()      { 
  263. 			console.log ("setRemoteDescription ok");
  264. 			while (bind.earlyCandidates.length) {
  265. 				var candidate = bind.earlyCandidates.shift();
  266. 				bind.addIceCandidate.call(bind, bind.pc.peerid, candidate);				
  267. 			}
  268. 		
  269. 			bind.getIceCandidate.call(bind)
  270. 		}
  271. 		, function(error) { 
  272. 			console.log ("setRemoteDescription error:" + JSON.stringify(error)); 
  273. 		});
  274. }	
  275.  
  276. /*
  277. * AJAX /getIceCandidate callback
  278. */
  279. WebRtcStreamer.prototype.onReceiveCandidate = function(dataJson) {
  280. 	console.log("candidate: " + JSON.stringify(dataJson));
  281. 	if (dataJson) {
  282. 		for (var i=0; i<dataJson.length; i++) {
  283. 			var candidate = new RTCIceCandidate(dataJson[i]);
  284. 			
  285. 			console.log("Adding ICE candidate :" + JSON.stringify(candidate) );
  286. 			this.pc.addIceCandidate(candidate
  287. 				, function()      { console.log ("addIceCandidate OK"); }
  288. 				, function(error) { console.log ("addIceCandidate error:" + JSON.stringify(error)); } );
  289. 		}
  290. 		this.pc.addIceCandidate();
  291. 	}
  292. }
  293.  
  294.  
  295. /*
  296. * AJAX callback for Error
  297. */
  298. WebRtcStreamer.prototype.onError = function(status) {
  299. 	console.log("onError:" + status);
  300. }
  301.  
  302. return WebRtcStreamer;
  303. })();
  304.  
  305. if (typeof module !== 'undefined' && typeof module.exports !== 'undefined')
  306. 	module.exports = WebRtcStreamer;
  307. else
  308. 	window.WebRtcStreamer = WebRtcStreamer;

这里启用需要下载webRTC

  https://github.com/mpromonet/webrtc-streamer/releases

需要注意的是这里启动不要直接双击而是使用cmd命令启动

 

start 应用名 -o 

一定加上-o否则webRTC占cpu太大 容易卡死


解决卡花屏问题:
在html页面中的webRtcServer.connect(s.data,"","rtptransport=tcp");加上"","rtptransport=tcp"就搞定
  1. <html>
  2. 	<head>
  3. 		<script src="js/webrtcstreamer.js"></script>
  4. 		<script>
  5. 			// 接受从vue组件中传过来的参数
  6. 			let url = location.search; //这一条语句获取了包括问号开始到参数的最后，不包括前面的路径
  7. 			let params = url.substr(1); //去掉问号
  8. 			let pa = params.split("&");
  9. 			let s = new Object();
  10.              //  设置后端服务地址
  11.             let VIDEOURL = "http://172.18.127.7:8000" //服务视频webrtc
  12.  
  13. 			for (let i = 0; i < pa.length; i++) {
  14. 				s[pa[i].split("=")[0]] = unescape(pa[i].split("=")[1]);
  15. 			}
  16. 			console.log(s.data)
  17. 			window.onload = function() {
  18. 				webRtcServer = new WebRtcStreamer("video", VIDEOURL);
  19. 				webRtcServer.connect(s.data,"","rtptransport=tcp");
  20. 			}
  21. 			window.onbeforeunload = function() {
  22. 				webRtcServer.disconnect();
  23. 			}
  24. 		</script>
  25. 	</head>
  26. 	<body>
  27. 		<h1 value="da3"></h1>
  28. 		<video id="video" style="width: 100%;height: 100%;" controls autoplay muted />
  29. 	</body>
  30. </html>
  31.

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