WEBRTC 视频直播记录

你在看桥下风景
  • 需求:解决RTMP/HLS/FLV 视频直播流延迟
  • 背景:因为视频直播是VR视频直播,直播流是属于8K 4K,而VR直播流比平常的平面直播要大特别多,所以在网络分发(CDN)中会有存在延迟,及H5播放中因为网络问题存在的流缓存导致延迟增大,特别是hls及flv播放模式,网络问题越大导致的流延迟越高
  • 优化目标:延迟在1s-2s(除去相机本身延迟)
  • 优化方案:WEBRTC + SRS 服务
  • 链接:
    SRS:https://github.com/ossrs/srs (身为前端的我并看不懂)
    WEBRTC:https://juejin.cn/post/684490... 简单介绍
    SRSWebRTCDemo:http://ossrs.net/srs.release/... SRSwebrtc演示

JS 资源
https://ossrs.net/players/js/...
https://ossrs.net/players/js/...
https://ossrs.net/players/js/...


  • 实时:主要说WEBRTC方法,
  •   <template>
            <div>
                <video id="rtc_media_player" autoplay></video>
                <!-- <video id="rtc_media_player" x-webkit-airplay='allow' webkit-playsinline playsinline controls
                x5-video-player-type='h5' x5-video-player-fullscreen x5-video-orientation='portrait' crossOrigin='Anonymous'
                allowsInlineMediaPlayback='true' autoplay></video> -->
            </div>
        </template>
    

提供一个Video的标签,注释里面有他的一些属性,srs播放需要一个video的ID rtc_media_player

    $(function () {
        // Async-await-promise based SRS RTC Player.
        function SrsRtcPlayerAsync() {
            var self = {};
            self.play = async function (url) {
                var conf = self.__internal.prepareUrl(url);
                self.pc.addTransceiver("audio", { direction: "recvonly" });
                self.pc.addTransceiver("video", { direction: "recvonly" });

                var offer = await self.pc.createOffer();
                await self.pc.setLocalDescription(offer);
                var session = await new Promise(function (resolve, reject) {
                    // @see https://github.com/rtcdn/rtcdn-draft
                    var data = {
                        api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
                    };
                    console.log("Generated offer: ", data);

                    $.ajax({
                        type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
                        contentType: 'application/json', dataType: 'json'
                    }).done(function (data) {
                        console.log("Got answer: ", data);
                        if (data.code) {
                            reject(data); return;
                        }

                        resolve(data);
                    }).fail(function (reason) {
                        reject(reason);
                    });
                });
                console.log(session)
                await self.pc.setRemoteDescription(
                    new RTCSessionDescription({ type: 'answer', sdp: session.sdp })
                );
                return session;
            };

            // Close the publisher.
            self.close = function () {
                self.pc.close();
            };

            // The callback when got remote stream.
            self.onaddstream = function (event) { };

            // Internal APIs.
            self.__internal = {
                defaultPath: '/rtc/v1/play/',
                prepareUrl: function (webrtcUrl) {
                    var urlObject = self.__internal.parse(webrtcUrl);

                    // If user specifies the schema, use it as API schema.
                    var schema = urlObject.user_query.schema;
                    schema = schema ? schema + ':' : window.location.protocol;

                    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 || self.__internal.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 };
                },
                parse: function (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
                    };
                    self.__internal.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;
                },
                fill_query: function (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;
                    }
                }
            };

            self.pc = new RTCPeerConnection(null);
            self.pc.onaddstream = function (event) {
                if (self.onaddstream) {
                    self.onaddstream(event);
                }
            };

            return self;
        }
        var sdk = null; // Global handler to do cleanup when replaying.
        var startPlay = function () {
            $('#rtc_media_player').show();

            // Close PC when user replay.
            if (sdk) {
                sdk.close();
            }

            sdk = new SrsRtcPlayerAsync();
            sdk.onaddstream = function (event) {
                console.log('Start play, event: ', event);
                console.log(event.stream)
                $('#rtc_media_player').prop('srcObject', event.stream);
            };
            // For example:
            //      webrtc://r.ossrs.net/live/livestream
            var url = $("#txt_url").val();
            sdk.play(url).then(function (session) {
                $('#sessionid').html(session.sessionid);
                $('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
            }).catch(function (reason) {
                sdk.close();
                $('#rtc_media_player').hide();
                console.error(reason);
            });
        };

        $('#rtc_media_player').hide();
        var query = parse_query_string();

        srs_init_rtc("#txt_url", query);
        $("#txt_url").val('webrtc://47.115.33.66/live/livestream')
        $("#btn_play").click(function () {
            $('#rtc_media_player').prop('muted', false);
            startPlay();
            setTimeout(() => {
                // vrVideoinit()
            }, 1000);
        });

        if (query.autostart === 'true') {
            $('#rtc_media_player').prop('muted', true);
            console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
                'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');

            startPlay();
        }
    });
    function vrVideoinit() {
        var scene, renderer;
        var container;
        //renderer = new THREE.WebGLRenderer();
        AVR.debug = true;
        if (!AVR.Broswer.isIE() && AVR.Broswer.webglAvailable()) {
            renderer = new THREE.WebGLRenderer();
        } else {
            renderer = new THREE.CanvasRenderer();
        }
        renderer.setPixelRatio(window.devicePixelRatio);
        container = document.getElementById('example');
        container.appendChild(renderer.domElement);
        scene = new THREE.Scene();
        // fov 选项可调整初始视频远近
        var vr = new VR(scene, renderer, container, { "fov": 90 });
        //vr.playText="<img src='img/play90.png' width='40' height='40'/>";
        vr.vrbox.radius = 600;
        if (AVR.isCrossScreen()) {
            // 调整vr视窗偏移量
            vr.effect.separation = 1.2;
        }
        vr.loadProgressManager.onLoad = function () {
            // 视频静音
            vr.video.muted = true;
        }
        //AVR.useGyroscope=false;
        vr.init(function () {

        });
        var videoDom = document.getElementById('rtc_media_player')
        vr.play(videoDom, vr.resType.webrtcVideo);
        vr.video.addEventListener('canplay', function () {
            vr.video.play()
        })
        vr.video.crossOrigin = "Anonymous";
        vr.video.onended = function () {
            console.log('结束?')
            $("#example").hide()
            $(".shade").show()
            $("#iframeDoms").show()
        }
    }
    

SrsRtcPlayerAsync 里面是SRS 提供内置RTC方法,及RTC解析,划重点

            self.play = async function (url) {
                var conf = self.__internal.prepareUrl(url);
                self.pc.addTransceiver("audio", { direction: "recvonly" });
                self.pc.addTransceiver("video", { direction: "recvonly" });

                var offer = await self.pc.createOffer();
                await self.pc.setLocalDescription(offer);
                var session = await new Promise(function (resolve, reject) {
                    // @see https://github.com/rtcdn/rtcdn-draft
                    var data = {
                        api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
                    };
                    console.log("Generated offer: ", data);
                    $.ajax({
                        type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
                        contentType: 'application/json', dataType: 'json'
                    }).done(function (data) {
                        console.log("Got answer: ", data);
                        if (data.code) {
                            reject(data); return;
                        }

                        resolve(data);
                    }).fail(function (reason) {
                        reject(reason);
                    });
                });
                console.log(session)
                await self.pc.setRemoteDescription(
                    new RTCSessionDescription({ type: 'answer', sdp: session.sdp })
                );
                return session;
            };


注意play方法中的 apiUrl,这个地址是需要后台提供支持的地址,也就是SRS服务中提供的WEBRTC 地址,会返回一些内置方法的参数

                $('#rtc_media_player').prop('srcObject', event.stream);

这个简单 给video 注入media 的Object,vrVideoinit方法是VR全景播放的方法,可忽略,部分代码提供

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