基于 uniapp 开发 android 播放 webrtc 流

一、播放rtsp协议流
如果 webrtc 流以 rtsp 协议返回，流地址如：rtsp://127.0.0.1:5115/session.mpg，uniapp的 <video> 编译到android上直接就能播放，但通常会有2-3秒的延迟。

二、播放webrtc协议流
如果 webrtc 流以 webrtc 协议返回，流地址如：webrtc://127.0.0.1:1988/live/livestream，我们需要通过sdp协商、连接推流服务端、搭建音视频流通道来播放音视频流，通常有500毫秒左右的延迟。

封装 WebrtcVideo 组件

<template>
	<video id="rtc_media_player" width="100%" height="100%" autoplay playsinline></video>
</template>

<!-- 因为我们使用到 js 库，所以需要使用 uniapp 的 renderjs -->
<script module="webrtcVideo" lang="renderjs">
	import $ from "./jquery-1.10.2.min.js";
	import {prepareUrl} from "./utils.js";
	
	export default {
	    data() {
	        return {
	        	//RTCPeerConnection 对象
	            peerConnection: null,
	            //需要播放的webrtc流地址
	            playUrl: 'webrtc://127.0.0.1:1988/live/livestream'
	        }
	    },
	    methods: {
	          createPeerConnection() {
		      	const that = this
		      	//创建 WebRTC 通信通道
	            that.peerConnection = new RTCPeerConnection(null);
	            //添加一个单向的音视频流收发器
				that.peerConnection.addTransceiver("audio", { direction: "recvonly" });
				that.peerConnection.addTransceiver("video", { direction: "recvonly" });
				//收到服务器码流，将音视频流写入播放器
	            that.peerConnection.ontrack = (event) => {
	                const remoteVideo = document.getElementById("rtc_media_player");
	                if (remoteVideo.srcObject !== event.streams[0]) {
	                    remoteVideo.srcObject = event.streams[0];
	                }
	            };
	        },
	        async makeCall() {
				const that = this
		        const url = this.playUrl
	            this.createPeerConnection()
	            //拼接服务端请求地址，如：http://192.168.0.1:1988/rtc/v1/play/
	            const conf = prepareUrl(url);
	            //生成 offer sdp
	            const offer = await this.peerConnection.createOffer();
	            await this.peerConnection.setLocalDescription(offer);
	            var session = await new Promise(function (resolve, reject) {
		            $.ajax({
		               type: "POST",
		               url: conf.apiUrl,
		               data: offer.sdp,
		               contentType: "text/plain",
		               dataType: "json",
		               crossDomain: true,
		           })
		           .done(function (data) {
		           	   //服务端返回 answer sdp
		               if (data.code) {
							reject(data);
							return;
		            	}
		                resolve(data);
		            })
		            .fail(function (reason) {
						reject(reason);
		            });
	            });
	            //设置远端的描述信息，协商sdp，通过后搭建通道成功
	            await this.peerConnection.setRemoteDescription(
	           		new RTCSessionDescription({ type: "answer", sdp: session.sdp })
	         	);
				session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'
				return session;
	        }
	    },
	       mounted() {
	           try {
				this.makeCall().then((res) => {
					// webrtc 通道建立成功
				})
			   } catch (error) {
				   // webrtc 通道建立失败
				   console.log(error)
			   }
	       }
	}
</script>

utils.js

const defaultPath = "/rtc/v1/play/";

export const prepareUrl = webrtcUrl => {
	var urlObject = parseUrl(webrtcUrl);
	var schema = "http:";
	var port = urlObject.port || 1985;
	if (schema === "https:") {
		port = urlObject.port || 443;
	}

	// @see https://github.com/rtcdn/rtcdn-draft
	var api = urlObject.user_query.play || defaultPath;
	if (api.lastIndexOf("/") !== api.length - 1) {
		api += "/";
	}

	apiUrl = schema + "//" + urlObject.server + ":" + port + api;
	for (var key in urlObject.user_query) {
		if (key !== "api" && key !== "play") {
			apiUrl += "&" + key + "=" + urlObject.user_query[key];
		}
	}
	// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
	var apiUrl = apiUrl.replace(api + "&", api + "?");

	var streamUrl = urlObject.url;

	return {
		apiUrl: apiUrl,
		streamUrl: streamUrl,
		schema: schema,
		urlObject: urlObject,
		port: port,
		tid: Number(parseInt(new Date().getTime() * Math.random() * 100))
			.toString(16)
			.substr(0, 7)
	};
};
export const parseUrl = url => {
	// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
	var a = document.createElement("a");
	a.href = url
		.replace("rtmp://", "http://")
		.replace("webrtc://", "http://")
		.replace("rtc://", "http://");

	var vhost = a.hostname;
	var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
	var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);

	// parse the vhost in the params of app, that srs supports.
	app = app.replace("...vhost...", "?vhost=");
	if (app.indexOf("?") >= 0) {
		var params = app.substr(app.indexOf("?"));
		app = app.substr(0, app.indexOf("?"));

		if (params.indexOf("vhost=") > 0) {
			vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
			if (vhost.indexOf("&") > 0) {
				vhost = vhost.substr(0, vhost.indexOf("&"));
			}
		}
	}

	// when vhost equals to server, and server is ip,
	// the vhost is __defaultVhost__
	if (a.hostname === vhost) {
		var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
		if (re.test(a.hostname)) {
			vhost = "__defaultVhost__";
		}
	}

	// parse the schema
	var schema = "rtmp";
	if (url.indexOf("://") > 0) {
		schema = url.substr(0, url.indexOf("://"));
	}

	var port = a.port;
	if (!port) {
		if (schema === "http") {
			port = 80;
		} else if (schema === "https") {
			port = 443;
		} else if (schema === "rtmp") {
			port = 1935;
		}
	}

	var ret = {
		url: url,
		schema: schema,
		server: a.hostname,
		port: port,
		vhost: vhost,
		app: app,
		stream: stream
	};
	fill_query(a.search, ret);

	// For webrtc API, we use 443 if page is https, or schema specified it.
	if (!ret.port) {
		if (schema === "webrtc" || schema === "rtc") {
			if (ret.user_query.schema === "https") {
				ret.port = 443;
			} else if (window.location.href.indexOf("https://") === 0) {
				ret.port = 443;
			} else {
				// For WebRTC, SRS use 1985 as default API port.
				ret.port = 1985;
			}
		}
	}

	return ret;
};
export const fill_query = (query_string, obj) => {
	// pure user query object.
	obj.user_query = {};

	if (query_string.length === 0) {
		return;
	}

	// split again for angularjs.
	if (query_string.indexOf("?") >= 0) {
		query_string = query_string.split("?")[1];
	}

	var queries = query_string.split("&");
	for (var i = 0; i < queries.length; i++) {
		var elem = queries[i];

		var query = elem.split("=");
		obj[query[0]] = query[1];
		obj.user_query[query[0]] = query[1];
	}

	// alias domain for vhost.
	if (obj.domain) {
		obj.vhost = obj.domain;
	}
};

页面中使用

<template>
	<VideoWebrtc />
</template>
<script setup>
	import VideoWebrtc from "@/components/videoWebrtc";
</script>

需要注意的事项：
1.spd 协商的重要标识之一为媒体描述： m=xxx <type> <code>，示例行如下：

一个完整的媒体描述，从第一个m=xxx <type> <code>开始，到下一个m=xxx <type> <code>结束，以video为例，媒体描述包含了当前设备允许播放的视频流编码格式，常见如：VP8/VP9/H264 等：

对照 m=video 后边的编码发现，其包含所有 a=rtpmap 后的编码，a=rtpmap 编码后的字符串代表视频流格式，但视频编码与视频流格式却不是固定的匹配关系，也就是说，在设备A中，可能存在 a=rtpmap:106 H264/90000 表示h264，在设备B中，a=rtpmap:100 H264/90000 表示h264。

因此，如果要鉴别设备允许播放的视频流格式，我们需要观察 a=rtpmap code 后的字符串。

协商通过的部分标准为：

1. offer sdp 的 m=xxx 数量需要与 answer sdp 的 m=xxx 数量保持一致；
2. offer sdp 的 m=xxx 顺序需要与 answer sdp 的 m=xxx 顺序保持一致；如两者都需要将 m=audio 放在第一位，m=video放在第二位，或者反过来；
3. answer sdp 返回的 m=audio 后的 <code>，需要被包含在 offer sdp 的 m=audio 后的<code>中；

offer sdp 的 m=xxx 由 addTransceiver 创建，首个参数为 audio 时，生成 m=audio，首个参数为video时，生成 m=video ，创建顺序对应 m=xxx 顺序

"recvonly" }); that.peerConnection.addTransceiver("video", {
direction: "recvonly" }); ```

4. 在 sdp 中存在一项 a=mid:xxx xxx在浏览器中可能为 audio、video ，在 android 设备上为 0、1，服务端需注意与 offer sdp 匹配。
5. 关于音视频流收发器，上面使用的api是 addTransceiver ，但在部分android设备上会提示没有这个api，我们可以替换为 getUserMedia + addTrack：

data() {
	return {
		......
	    localStream: null,
	    ......
	}
},
methods: {
	createPeerConnection() {
		const that = this
		//创建 WebRTC 通信通道
	    that.peerConnection = new RTCPeerConnection(null);
	    that.localStream.getTracks().forEach((track) => {
          that.peerConnection.addTrack(track, that.localStream);
        });
        //收到服务器码流，将音视频流写入播放器
	    that.peerConnection.ontrack = (event) => {
	    	......
	    };
	}，    
	async makeCall() {
		const that = this
		that.localStream = await navigator.mediaDevices.getUserMedia({
	    	video: true,
	        audio: true,
	    });
		const url = this.playUrl
		......
		......
	}
}

需要注意的是，navigator.mediaDevices.getUserMedia
获取的是设备摄像头、录音的媒体流，所以设备首先要具备摄像、录音功能，并开启对应权限，否则 api 将调用失败。

三、音视频实时通讯
这种 p2p 场景的流播放，通常需要使用 websocket 建立服务器连接，然后同时播放本地、服务端的流。

<template>
	<div>Local Video</div>
	<video id="localVideo" autoplay playsinline></video>
	<div>Remote Video</div>
	<video id="remoteVideo" autoplay playsinline></video>
</template>
<script module="webrtcVideo" lang="renderjs">
	import $ from "./jquery-1.10.2.min.js";
	export default {
		data() {
	        return {
	            signalingServerUrl: "ws://127.0.0.1:8085",
	            iceServersUrl: 'stun:stun.l.google.com:19302',
	            localStream: null,
	            peerConnection: null
	        }
	    },
	     methods: {
	     	async startLocalStream(){
		     	try {
		     		this.localStream = await navigator.mediaDevices.getUserMedia({
            			video: true,
            			audio: true,
          			});
          			document.getElementById("localVideo").srcObject = this.localStream;
		     	}catch (err) {
		     		console.error("Error accessing media devices.", err);
		     	}
	     	},
	     	createPeerConnection() {
	     		const configuration = { iceServers: [{ 
	     			urls: this.iceServersUrl 
	     		}]};
	     		this.peerConnection = new RTCPeerConnection(configuration);
	     		this.localStream.getTracks().forEach((track) => {
          			this.peerConnection.addTrack(track, this.localStream);
        		});
        		this.peerConnection.onicecandidate = (event) => {
          			if (event.candidate) {
			            ws.send(
			              JSON.stringify({
			                type: "candidate",
			                candidate: event.candidate,
			              })
			            );
			         }
        		};
        		this.peerConnection.ontrack = (event) => {
		          const remoteVideo = document.getElementById("remoteVideo");
		          if (remoteVideo.srcObject !== event.streams[0]) {
		            remoteVideo.srcObject = event.streams[0];
		          }
        		};
	     	}，
	     	async makeCall() {
	     		this.createPeerConnection();
	     		const offer = await this.peerConnection.createOffer();
	     		await this.peerConnection.setLocalDescription(offer);
	     		ws.send(JSON.stringify(offer));
	     	}
	     },
	     mounted() {
	     	this.makeCall()
	     	const ws = new WebSocket(this.signalingServerUrl);
	     	 ws.onopen = () => {
		     	console.log("Connected to the signaling server");
		        this.startLocalStream();
      		};
      		ws.onmessage = async (message) => {
      			const data = JSON.parse(message.data);
      			if (data.type === "offer") {
          			if (!this.peerConnection) createPeerConnection();
          			await this.peerConnection.setRemoteDescription(
            			new RTCSessionDescription(data)
          			);
          			const answer = await this.peerConnection.createAnswer();
          			await this.peerConnection.setLocalDescription(answer);
          			ws.send(JSON.stringify(this.peerConnection.localDescription));
        		} else if (data.type === "answer") {
          			if (!this.peerConnection) createPeerConnection();
          				await this.peerConnection.setRemoteDescription(
            				new RTCSessionDescription(data)
          				);
        			} else if (data.type === "candidate") {
          				if (this.peerConnection) {
				            try {
				              await this.peerConnection.addIceCandidate(
				                new RTCIceCandidate(data.candidate)
				              );
				            } catch (e) {
				              console.error("Error adding received ICE candidate", e);
				            }
          				}
        			}
      			}
	     }
	}
</script>

与播放webrtc协议流相比，p2p 以 WebSocket 替代 ajax 实现 sdp 的发送与接收，增加了本地流的播放功能，其他与播放协议流的代码一致。