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 = thisconst url = this.playUrlthis.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 sdpconst 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 sdpif (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-draftvar 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=vvar 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-javascrivar 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 schemavar 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 = thisthat.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。