Gahing's blog
gahing
2016/08/24
目录

流媒体直播—WebRTC native 2 browser

# 前言

关于browser 2 browser的网上文章很多，我就不介绍了。

可以参考我的这篇文章流媒体直播——资料整理、你想要了解的都在这 (opens new window)

里面给的链接可以去看看

这边我要讲的是WebRTC编译后的执行文件和Web端进行交互。

首先要明确的是，browser 2 browser的应用场景太狭小了：视频通话...

而编译WebRTC我们就可以做更多的事：屏幕共享.. 录播.. 远程教育【切换通话和屏幕】..

因为视频源我们是可控的，不像 Web. 【其实b2b也可以做屏幕共享，不过c2b会比较可控】

编译WebRTC是一件麻烦的事，可以参考上面那篇文章。 这里我直接给出两个可运行程序

下载地址：http://pan.baidu.com/s/1bVpx9O (opens new window)

如果网盘挂了请提醒我

# 流程

先用起来吧，虽然js有注释，不过我后面还是会详细解析的。

# 构架Web端

  1. 新建一个文件夹作为该web项目的根目录，就叫WebrtcTEST吧

  2. 在根目录下新建js文件夹

  3. 下载所需js文件.rar (opens new window)并解压，把得到的文件放入js文件夹中。

  4. 根目录下新建test.html文件，代码如下

    <!DOCTYPE html> 
     <head>
 	<meta charset="utf-8" />
 	<title>游戏</title>
 	<script type="text/javascript" src="./jquery-1.11.1.min.js"></script>
 	<script type="text/javascript" src="./adapter.js"></script>
 	<!--<script type="text/javascript" src="./adapter_no_global.js"></script>-->
 	<script>
 		var request = null;
 		var hangingGet = null;
 		var localName;
 		var server;
 		var my_id = -1;
 		//记录其他peer
 		var other_peers = {};
 		var message_counter = 0;

 		// added stuff start 
 		var pc;
 		//constraints for desktop browser 
 		var desktopConstraints = {

 			video: {
 				mandatory: {
 					maxWidth: 800,
 					maxHeight: 600
 				}
 			},

 			audio: true
 		};

 		//constraints for mobile browser 
 		var mobileConstraints = {

 			video: {
 				mandatory: {
 					maxWidth: 480,
 					maxHeight: 320,
 				}
 			},

 			audio: true
 		}

 		//if a user is using a mobile browser 
 		if(/Android|iPhone|iPad/i.test(navigator.userAgent)) {
 			var constraints = mobileConstraints;
 		} else {
 			var constraints = desktopConstraints;
 		}
 		var remoteStream;

 		function trace(txt) {
 			var elem = document.getElementById("debug");
 			elem.innerHTML += txt + "<br>";
 		}

 		function sendToPeer(peer_id, data) {
 			if(my_id == -1) {
 				alert("Not connected");
 				return;
 			}
 			if(peer_id == my_id) {
 				alert("Can't send a message to oneself :)");
 				return;
 			}
 			var r = new XMLHttpRequest();
 			r.open("POST", server + "/message?peer_id=" + my_id + "&to=" + peer_id,
 				false);
 			r.setRequestHeader("Content-Type", "text/plain");
 			r.send(data);
 			console.log(peer_id, " Send ", data);
 			trace(peer_id + " Send :" + data);
 			var dataJson = JSON.parse(data);
 			console.log(peer_id, " send ", data);
 			r = null;
 		}
 		var onSessionConnecting = function(message) {

 			console.log("Session connecting.");
 			trace("Session connecting.");
 		}

 		var onSessionOpened = function(message) {
 			console.log("Session opened.");
 			trace("Session opened.");
 		}

 		var onRemoteStreamRemoved = function(event) {
 				console.log("Remote stream removed.");
 				trace("Remote stream removed.");
 			}
 			//This function 
 		var createPeerConnection = function(connectionId) {
 			var pc_config = {
 				"iceServers": [{
 					"url": "stun:stun.l.google.com:19302"
 				}]
 			};
 			try {
 				console.log(pc_config);
 				trace(pc_config);
 				// 创建PeerConnection实例 (参数为null则没有iceserver，即使没有stunserver和turnserver，仍可在局域网下通讯)
 				pc = new RTCPeerConnection(pc_config);
 				// 发送ICE候选到其他客户端
 				pc.onicecandidate = function(event) {
 					if(event.candidate) {
 						var msgCANDIDATE = {};
 						msgCANDIDATE.sdpMLineIndex = event.candidate.sdpMLineIndex;
 						msgCANDIDATE.sdpMid = event.candidate.sdpMid;
 						msgCANDIDATE.candidate = event.candidate.candidate;
 						sendToPeer(connectionId, JSON.stringify(msgCANDIDATE));
 					} else {
 						console.log("End of candidates.");
 						trace("End of candidates.");
 					}
 				};
 				pc.onconnecting = onSessionConnecting;
 				pc.onopen = onSessionOpened;
 				// 如果检测到媒体流连接到本地，将其绑定到一个video标签上输出
 				pc.onaddstream = function(event) {
 					console.log("Remote stream added.");
 					trace("Remote stream added.");
 					document.getElementById('remoteVideo').src = URL.createObjectURL(event.stream);
 				}
 				pc.onremovestream = onRemoteStreamRemoved;
 				console.log("Created RTCPeerConnnection with config \"" + JSON.stringify(pc_config) + "\". for ");
 				trace("Created RTCPeerConnnection with config \"" + JSON.stringify(pc_config) + "\". for ");
 				//setLocalStream(connectionId);//解除注释将可以双端通信 不过延迟就高了
 			} catch(e) {
 				console.log("Failed to create PeerConnection with " + connectionId + ", exception: " + e.message);
 				trace("Failed to create PeerConnection with " + connectionId + ", exception: " + e.message);
 			}

 		}

 		function setLocalStream(connectionId) {
 			// 获取本地音频和视频流
 			navigator.getUserMedia(constraints, function(stream) {
 				//绑定本地媒体流到video标签用于输出
 				document.getElementById('localVideo').src = URL.createObjectURL(stream);
 				//向PeerConnection中加入需要发送的流
 				pc.addStream(stream);
 				//如果是发起方则发送一个offer信令
 				pc.createOffer(function sendOfferFn(desc) {
 					pc.setLocalDescription(desc);
 					var data = JSON.stringify(desc);
 					sendToPeer(connectionId, data);
 				}, function(error) {
 					console.log('Failure callback: ' + error);
 					trace('Failure callback: ' + error);
 				});
 			}, function(error) {
 				//处理媒体流创建失败错误
 				console.log('getUserMedia error: ' + error);
 				trace('getUserMedia error: ' + error);
 			});

 		}
 		//添加新加入的peer到本地
 		function handleServerNotification(data) {
 			console.log("Server notification: " + data);
 			trace("Server notification: " + data);
 			var parsed = data.split(',');
 			if(parseInt(parsed[2]) != 0)
 				other_peers[parseInt(parsed[1])] = parsed[0];
 		}
 		//如果是一个ICE的候选，则将其加入到PeerConnection中，否则设定对方的session描述为传递过来的描述
 		function handlePeerMessage(peer_id, data) {
 			var dataJson = JSON.parse(data);
 			if(data.search("offer") != -1) {
 				//json:{"sdp":xxx,type:"offer"}
 				createPeerConnection(peer_id);
 				pc.setRemoteDescription(new RTCSessionDescription(dataJson));
 				pc.createAnswer(function(sessionDescription) {
 					var answerData = JSON.stringify(sessionDescription);
 					console.log("answer:" + answerData);
 					trace("answer:" + answerData);
 					pc.setLocalDescription(sessionDescription);
 					sendToPeer(peer_id, answerData);
 				}, function(error) {
 					console.log('Failure callback: ' + error);
 					trace('Failure callback: ' + error);
 				});
 			} else {
 				//It is a candidate
 				var candidate = new RTCIceCandidate({
 					sdpMLineIndex: dataJson.sdpMLineIndex,
 					candidate: dataJson.candidate
 				});
 				pc.addIceCandidate(candidate);
 			}
 		}

 		function GetIntHeader(r, name) {
 			var val = r.getResponseHeader(name);
 			return val != null && val.length ? parseInt(val) : -1;
 		}
 		//等待服务器返回，当出现以下情况或超时才会返回 
 		//1.其他peer连接上server会返回peer的数据{peerName,peerID,xx};用于本地添加
 		//本地响应handleServerNotification
 		//2.其他peer点击my_name(即发送 发送-接收视频 请求)，返回 一个ice的候选或者offer
 		//	{
 		// "candidate" : "candidate:1918330882 1 udp 2122260223 192.168.253.1 55710 typ host generation 0 ufrag YUtt network-id 4 network-cost 50",
 		// "sdpMLineIndex" : 0,
 		// "sdpMid" : "audio"
 		//} this is ice
 		//本地响应handlePeerMessage
 		function hangingGetCallback() {
 			try {
 				if(hangingGet.readyState != 4)
 					return; //why?
 				if(hangingGet.status != 200) {
 					console.log("server error: " + hangingGet.statusText);
 					trace("server error: " + hangingGet.statusText);
 					disconnect();
 				} else {
 					var peer_id = GetIntHeader(hangingGet, "Pragma");

 					if(peer_id == my_id) {
 						console.log("handleServerNotification:" + hangingGet.responseText);
 						trace("handleServerNotification:" + hangingGet.responseText);
 						handleServerNotification(hangingGet.responseText);
 					} else {
 						console.log("handlePeerMessage:" + hangingGet.responseText);
 						trace("handlePeerMessage:" + hangingGet.responseText);
 						handlePeerMessage(peer_id, hangingGet.responseText);
 					}
 				}

 				if(hangingGet) {
 					hangingGet.abort();
 					hangingGet = null;
 				}

 				if(my_id != -1)
 					window.setTimeout(startHangingGet, 0);
 			} catch(e) {
 				console.log("Hanging get error: " + e.description);
 				trace("Hanging get error: " + e.description);
 			}
 		}
 		//继续把my_id发给server 当其他peer连接服务器的时候其他peer就可以获取my_id
 		function startHangingGet() {
 			try {
 				hangingGet = new XMLHttpRequest();
 				hangingGet.onreadystatechange = hangingGetCallback;
 				hangingGet.ontimeout = onHangingGetTimeout;
 				hangingGet.open("GET", server + "/wait?peer_id=" + my_id, true);
 				hangingGet.send();
 			} catch(e) {
 				console.log("error" + e.description);
 				trace("error" + e.description);
 			}
 		}
 		//如果超时则继续发送startHangingGet请求
 		function onHangingGetTimeout() {
 			console.log("hanging get timeout. issuing again.");
 			trace("hanging get timeout. issuing again.");
 			hangingGet.abort();
 			hangingGet = null;
 			if(my_id != -1)
 				window.setTimeout(startHangingGet, 0);
 		}
 		//server返回my_id(服务器已递增的方式生成id)和远端peer的id
 		//那如果browser先连接，没有其他peer，该方法返回时没有远端peer的id 后面是哪里获取的呢？
 		//所以它又做了一个startHangingGet();
 		//通过handleServerNotification 得到,log:Server notification: zhengjx@X-CDN-zhengjx,2,1
 		function signInCallback() {
 			try {
 				console.log("request.readyState:" + request.readyState);
 				trace("request.readyState:" + request.readyState);
 				if(request.readyState == 4) {
 					if(request.status == 200) {
 						var peers = request.responseText.split("\n");
 						my_id = parseInt(peers[0].split(',')[1]);
 						console.log("My id: " + my_id); //My id 1
 						trace("My id: " + my_id); //My id 1
 						//Peer 1: zhengjx@X-CDN-zhengjx,2,parsed[2]没用到 
 						for(var i = 1; i < peers.length; ++i) {
 							if(peers[i].length > 0) {
 								console.log("Peer " + i + ": " + peers[i]);
 								trace("Peer " + i + ": " + peers[i]);
 								var parsed = peers[i].split(',');
 								other_peers[parseInt(parsed[1])] = parsed[0];
 							}
 						}
 						startHangingGet();
 						request = null;
 					}
 				}
 			} catch(e) {
 				console.log("error: " + e.description);
 				trace("error: " + e.description);
 			}
 		}
 		//注册，把自己的localName发给server
 		function signIn() {
 			try {
 				request = new XMLHttpRequest();
 				request.onreadystatechange = signInCallback;
 				request.open("GET", server + "/sign_in?" + localName, true);
 				request.send();
 			} catch(e) {
 				console.log("error: " + e.description);
 				trace("error: " + e.description);
 			}
 		}

 		function connect() {
 			localName = document.getElementById("local").value.toLowerCase();
 			server = document.getElementById("server").value.toLowerCase();
 			if(localName.length == 0) {
 				alert("I need a name please.");
 				document.getElementById("local").focus();
 			} else {
 				document.getElementById("connect").disabled = true;
 				document.getElementById("disconnect").disabled = false;
 				//document.getElementById("send").disabled = false;
 				signIn();
 			}
 		}

 		function disconnect() {
 			if(request) {
 				request.abort();
 				request = null;
 			}

 			if(hangingGet) {
 				hangingGet.abort();
 				hangingGet = null;
 			}

 			if(my_id != -1) {
 				request = new XMLHttpRequest();
 				request.open("GET", server + "/sign_out?peer_id=" + my_id, false);
 				request.send();
 				request = null;
 				my_id = -1;
 			}

 			document.getElementById("connect").disabled = false;
 			document.getElementById("disconnect").disabled = true;
 			//document.getElementById("send").disabled = true;
 		}
 	</script>
 </head>

 <body>
 	Local: <br>
 	<video id="localVideo" autoplay></video><br> Remote: <br>
 	<video id="remoteVideo" autoplay style="width:400px;height:300px;"></video>

 	Server: <input type="text" id="server" value="http://172.27.35.1:8888" /><br> Your name: <input type="text" id="local" value="my_name" />
 	<button id="connect" onclick="connect();">Connect</button>
 	<button disabled="true" id="disconnect" onclick="disconnect();">Disconnect</button>
 	<pre id="debug">
 	</pre>
 </body>

将WebrtcTEST项目放入HTTP服务器 访问 http://localhost/WebrtcTEST/test.html

# 服务器端

下载百度云的那个rar并解压得到 peerconnection_client.exe 和 peerconnection_server.exe peerconnection_client.exe的网络运行环境任意，可局域网可公网。 但peerconnection_server.exe网络运行环境必须保证能让peerconnection_client和Web端localhost所在网络运行环境访问到。 也就是说，要么peerconnection_server处于公网，要么三者均处于局域网。

PS:测试时为了方便选择局域网，并让peerconnection_client和peerconnection_server处于同一台PC

# 测试流程如下：

  1. 服务器端运行peerconnection_server

  2. 服务器端运行peerconnection_client，并直接点击connect

  3. 另一Client的Web端设置Server IP地址和Your name【name随意】,点击connect

  4. 服务器端peerconnection_client显示Web端连接上了，列表上出现name，对其进行点击;

  5. 现在就可以在Web端看到服务器端摄像头的画面了，延迟小于200ms.

  6. 如果想让服务器端看到Web端用户摄像头画面，将test.html的第132行注释去掉

    //setLocalStream(connectionId);//解除注释将可以双端通信 不过延迟就高了

# JS具体流程分析

# 从function connect() 开始分析

# 调用signIn()函数

将自己的localName【填写的Your name】发给server,响应结束将调用signInCallback

# 分析signInCallback()

server返回my_id(服务器已递增的方式生成id)和远端peer的id 那如果browser先连接，没有其他peer，该方法返回时没有远端peer的id 后面是哪里获取的呢？ 所以它又做了一个startHangingGet(); 返回的peer数据格式：peer_name,peer_id,第三个参数没有用到我也不知道是什么。

# 执行startHangingGet();

不管上一步有没有获取到peer该步骤都会进行的。

# 该步骤是把my_id发送给server.

当其他peer进行操作**[1.连接服务器的时候其他peer就可以获取my_id;2.发送播流指令]**XHR请求将成功响应,并且自己将执行hangingGetCallback。 如果请求超时的话将回调onHangingGetTimeout，其结果是重新执行startHangingGet

# 执行hangingGetCallback

正如刚刚说的，有两种情况，服务器将响应返回

# 1.其他peer连接上server会返回peer的数据{peerName,peerID,xx};用于本地添加

本地响应handleServerNotification log打印：

handleServerNotification:zhengjx@X-CDN-zhengjx,4,1
Server notification: zhengjx@X-CDN-zhengjx,4,1

# 2.其他peer点击my_name(即发送 播流 请求)，返回 一个ice的候选或者offer

本地响应handlePeerMessage

# 这里我们直接分析peer发送播流请求

# 远端peer先发送了一个携带音视频信息sdp的offer

handlePeerMessage:{
   "sdp" : "v=0\r\no=- 1915807044938492152 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS stream_label\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:s+rj\r\na=ice-pwd:JZspr0P/+RWnF3WFF7H2njf+\r\na=fingerprint:sha-256 DC:1B:45:26:86:B0:51:E1:2C:84:F6:04:83:1F:13:87:5F:54:9D:9F:4C:8D:04:33:E3:2F:69:BC:69:23:51:D5\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:102 ILBC/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:127 red/8000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:1946889360 cname:oeS68Tdb4Ls6Qz2u\r\na=ssrc:1946889360 msid:stream_label audio_label\r\na=ssrc:1946889360 mslabel:stream_label\r\na=ssrc:1946889360 label:audio_label\r\nm=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:s+rj\r\na=ice-pwd:JZspr0P/+RWnF3WFF7H2njf+\r\na=fingerprint:sha-256 DC:1B:45:26:86:B0:51:E1:2C:84:F6:04:83:1F:13:87:5F:54:9D:9F:4C:8D:04:33:E3:2F:69:BC:69:23:51:D5\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:100 VP8/90000\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtpmap:101 VP9/90000\r\na=rtcp-fb:101 ccm fir\r\na=rtcp-fb:101 nack\r\na=rtcp-fb:101 nack pli\r\na=rtcp-fb:101 goog-remb\r\na=rtcp-fb:101 transport-cc\r\na=rtpmap:116 red/90000\r\na=rtpmap:117 ulpfec/90000\r\na=rtpmap:96 rtx/90000\r\na=fmtp:96 apt=100\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=101\r\na=rtpmap:98 rtx/90000\r\na=fmtp:98 apt=116\r\na=ssrc-group:FID 1010645505 1298682012\r\na=ssrc:1010645505 cname:oeS68Tdb4Ls6Qz2u\r\na=ssrc:1010645505 msid:stream_label video_label\r\na=ssrc:1010645505 mslabel:stream_label\r\na=ssrc:1010645505 label:video_label\r\na=ssrc:1298682012 cname:oeS68Tdb4Ls6Qz2u\r\na=ssrc:1298682012 msid:stream_label video_label\r\na=ssrc:1298682012 mslabel:stream_label\r\na=ssrc:1298682012 label:video_label\r\n",
   "type" : "offer"
}

# createPeerConnection，设定对方的sdp为传递过来的描述

执行

createPeerConnection(peer_id);
pc.setRemoteDescription(new RTCSessionDescription(dataJson));				                    pc.createAnswer(function(sessionDescription)  {
		var answerData = JSON.stringify(sessionDescription);
		console.log("answer:" + answerData);
		trace("answer:" + answerData);
	    pc.setLocalDescription(sessionDescription);
		sendToPeer(peer_id, answerData);
					}, function(error) {
						console.log('Failure callback: ' + error);
						trace('Failure callback: ' + error);
					});

创建PeerConnection实例 (参数为null则没有iceserver，即使没有stunserver和turnserver，仍可在局域网下通讯) 这里我们设置了onicecandidate：

{"iceServers":[{"url":"stun:stun.l.google.com:19302"}]}

通过执行sendToPeer函数发送ICE候选到其他客户端，该过程会慢于RTCPeerConnection.onaddstream。

当上面做的pc.setRemoteDescription成功执行时，pc.onaddstream将调用。 官网原话:

RTCPeerConnection.onaddstream：
The event is sent immediately after the call RTCPeerConnection.setRemoteDescription()

回复的answer sdp 如下：

answer:{"type":"answer","sdp":"v=0\r\no=- 681837714082552550 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:XNSrC9DEJco2wMEV\r\na=ice-pwd:PhP5CUsS6P8wxnCZZa5aZ5nD\r\na=fingerprint:sha-256 F8:93:41:5C:50:E9:EF:11:66:0F:2F:64:F7:CE:C4:BB:C5:57:D8:3C:C3:04:19:27:33:B7:93:DE:58:9F:DF:74\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:126 telephone-event/8000\r\na=maxptime:60\r\nm=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:XNSrC9DEJco2wMEV\r\na=ice-pwd:PhP5CUsS6P8wxnCZZa5aZ5nD\r\na=fingerprint:sha-256 F8:93:41:5C:50:E9:EF:11:66:0F:2F:64:F7:CE:C4:BB:C5:57:D8:3C:C3:04:19:27:33:B7:93:DE:58:9F:DF:74\r\na=setup:active\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:100 VP8/90000\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtpmap:101 VP9/90000\r\na=rtcp-fb:101 ccm fir\r\na=rtcp-fb:101 nack\r\na=rtcp-fb:101 nack pli\r\na=rtcp-fb:101 goog-remb\r\na=rtcp-fb:101 transport-cc\r\na=rtpmap:116 red/90000\r\na=rtpmap:117 ulpfec/90000\r\na=rtpmap:96 rtx/90000\r\na=fmtp:96 apt=100\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=101\r\na=rtpmap:98 rtx/90000\r\na=fmtp:98 apt=116\r\n"}

# 接下来的过程

上一步提到的，本地发送ICE候选到其他客户端

" Send " "{"sdpMLineIndex":0,"sdpMid":"audio","candidate":"candidate:3604340130 1 udp 2122260223 192.168.216.1 50938 typ host generation 0 ufrag XNSrC9DEJco2wMEV network-id 4"}"

该过程会执行多次，每次携带的candidate都不一致，直到打印console.log("End of candidates."); 与此同时，本地通过handlePeerMessage获取到ICE的候选，并将其加入到PeerConnection中。 这样两端就都有各自的ICE候选了，那么就可以进行通信。

pc.onaddstream在前一步调用，那么接下来peer就应该开始传流。 本地通过handlePeerMessage获取的音视频数据部分如下

handlePeerMessage:{
   "candidate" : "candidate:2013099291 2 udp 2122194686 10.8.146.177 54148 typ host generation 0 ufrag s+rj network-id 3 network-cost 50",
   "sdpMLineIndex" : 0,
   "sdpMid" : "audio"
}

VM122 phone_frame.html:450 handlePeerMessage:{
   "candidate" : "candidate:2719042630 1 udp 2122260223 172.27.35.1 54149 typ host generation 0 ufrag s+rj network-id 4 network-cost 50",
   "sdpMLineIndex" : 1,
   "sdpMid" : "video"
}

[注意之间建立了的candidate会在传输时用到，所以可以理解candidate是一种管道，我们创建了多个candidate就是多个管道，视音频数据就是通过这些管道来传输] 当然，再具体的原理就不探讨了，涉及穿透等高深的知识= =。

由于研究未深，如有错误，烦请指正，谢谢。

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上次更新: 2025/06/11, 23:06:59
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