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增加新的安防接入方式

caoyang 6 months ago
parent
8cff6e680a
commit
290f94fae3
4 changed files with 750 additions and 2 deletions
Unified View Show Diff Stats
  1. 1 0
      index.html
  2. 681 0
      public/srs.sdk.js
  3. 2 2
      src/views/camera/index.vue
  4. 66 0
      src/views/camera/push.vue

+ 1 - 0
index.html
View File 

@@ -15,6 +15,7 @@
 
     />
 
     />
 
     <script src="/adapter.min.js"></script> 
     <script src="/adapter.min.js"></script> 
     <script src="/webrtcstreamer.js"></script> 
     <script src="/webrtcstreamer.js"></script> 
+    <script src="/srs.sdk.js"></script> 
     <title>%VITE_APP_TITLE%</title>
 
     <title>%VITE_APP_TITLE%</title>
 
   </head>
 
   </head>
 
   <body>
 
   <body>

+ 681 - 0
public/srs.sdk.js
View File 

@@ -0,0 +1,681 @@
 
+
 
+//
 
+// Copyright (c) 2013-2021 Winlin
 
+//
 
+// SPDX-License-Identifier: MIT
 
+//
 
+
 
+'use strict';
 
+
 
+function SrsError(name, message) {
 
+    this.name = name;
 
+    this.message = message;
 
+    this.stack = (new Error()).stack;
 
+}
 
+SrsError.prototype = Object.create(Error.prototype);
 
+SrsError.prototype.constructor = SrsError;
 
+
 
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
 
+// Async-awat-prmise based SRS RTC Publisher.
 
+function SrsRtcPublisherAsync() {
 
+    var self = {};
 
+
 
+    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
 
+    self.constraints = {
 
+        audio: true,
 
+        video: {
 
+            width: {ideal: 320, max: 576}
 
+        }
 
+    };
 
+
 
+    // @see https://github.com/rtcdn/rtcdn-draft
 
+    // @url The WebRTC url to play with, for example:
 
+    //      webrtc://r.ossrs.net/live/livestream
 
+    // or specifies the API port:
 
+    //      webrtc://r.ossrs.net:11985/live/livestream
 
+    // or autostart the publish:
 
+    //      webrtc://r.ossrs.net/live/livestream?autostart=true
 
+    // or change the app from live to myapp:
 
+    //      webrtc://r.ossrs.net:11985/myapp/livestream
 
+    // or change the stream from livestream to mystream:
 
+    //      webrtc://r.ossrs.net:11985/live/mystream
 
+    // or set the api server to myapi.domain.com:
 
+    //      webrtc://myapi.domain.com/live/livestream
 
+    // or set the candidate(eip) of answer:
 
+    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
 
+    // or force to access https API:
 
+    //      webrtc://r.ossrs.net/live/livestream?schema=https
 
+    // or use plaintext, without SRTP:
 
+    //      webrtc://r.ossrs.net/live/livestream?encrypt=false
 
+    // or any other information, will pass-by in the query:
 
+    //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
 
+    //      webrtc://r.ossrs.net/live/livestream?token=xxx
 
+    self.publish = async function (url) {
 
+        var conf = self.__internal.prepareUrl(url);
 
+        self.pc.addTransceiver("audio", {direction: "sendonly"});
 
+        self.pc.addTransceiver("video", {direction: "sendonly"});
 
+        //self.pc.addTransceiver("video", {direction: "sendonly"});
 
+        //self.pc.addTransceiver("audio", {direction: "sendonly"});
 
+
 
+        if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
 
+            throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
 
+        }
 
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
 
+
 
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
 
+        stream.getTracks().forEach(function (track) {
 
+            self.pc.addTrack(track);
 
+
 
+            // Notify about local track when stream is ok.
 
+            self.ontrack && self.ontrack({track: track});
 
+        });
 
+
 
+        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, tid: conf.tid, streamurl: conf.streamUrl,
 
+                clientip: null, sdp: offer.sdp
 
+            };
 
+            console.log("Generated offer: ", data);
 
+
 
+            const xhr = new XMLHttpRequest();
 
+            xhr.onload = function() {
 
+                if (xhr.readyState !== xhr.DONE) return;
 
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
 
+                const data = JSON.parse(xhr.responseText);
 
+                console.log("Got answer: ", data);
 
+                return data.code ? reject(xhr) : resolve(data);
 
+            }
 
+            xhr.open('POST', conf.apiUrl, true);
 
+            xhr.setRequestHeader('Content-type', 'application/json');
 
+            xhr.send(JSON.stringify(data));
 
+        });
 
+        await self.pc.setRemoteDescription(
 
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
 
+        );
 
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
 
+
 
+        return session;
 
+    };
 
+
 
+    // Close the publisher.
 
+    self.close = function () {
 
+        self.pc && self.pc.close();
 
+        self.pc = null;
 
+    };
 
+
 
+    // The callback when got local stream.
 
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
 
+    self.ontrack = function (event) {
 
+        // Add track to stream of SDK.
 
+        self.stream.addTrack(event.track);
 
+    };
 
+
 
+    // Internal APIs.
 
+    self.__internal = {
 
+        defaultPath: '/rtc/v1/publish/',
 
+        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 += '/';
 
+            }
 
+
 
+            var 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
 
+            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).slice(0, 7)
 
+            };
 
+        },
 
+        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.substring(1, a.pathname.lastIndexOf("/"));
 
+            var stream = a.pathname.slice(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.slice(app.indexOf("?"));
 
+                app = app.slice(0, app.indexOf("?"));
 
+
 
+                if (params.indexOf("vhost=") > 0) {
 
+                    vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
 
+                    if (vhost.indexOf("&") > 0) {
 
+                        vhost = vhost.slice(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.slice(0, url.indexOf("://"));
 
+            }
 
+
 
+            var port = a.port;
 
+            if (!port) {
 
+                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
 
+                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
 
+                    port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
 
+                }
 
+
 
+                // Guess by schema.
 
+                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);
 
+
 
+    // To keep api consistent between player and publisher.
 
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
 
+    // @see https://webrtc.org/getting-started/media-devices
 
+    self.stream = new MediaStream();
 
+
 
+    return self;
 
+}
 
+
 
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
 
+// Async-await-promise based SRS RTC Player.
 
+function SrsRtcPlayerAsync() {
 
+    var self = {};
 
+
 
+    // @see https://github.com/rtcdn/rtcdn-draft
 
+    // @url The WebRTC url to play with, for example:
 
+    //      webrtc://r.ossrs.net/live/livestream
 
+    // or specifies the API port:
 
+    //      webrtc://r.ossrs.net:11985/live/livestream
 
+    //      webrtc://r.ossrs.net:80/live/livestream
 
+    // or autostart the play:
 
+    //      webrtc://r.ossrs.net/live/livestream?autostart=true
 
+    // or change the app from live to myapp:
 
+    //      webrtc://r.ossrs.net:11985/myapp/livestream
 
+    // or change the stream from livestream to mystream:
 
+    //      webrtc://r.ossrs.net:11985/live/mystream
 
+    // or set the api server to myapi.domain.com:
 
+    //      webrtc://myapi.domain.com/live/livestream
 
+    // or set the candidate(eip) of answer:
 
+    //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
 
+    // or force to access https API:
 
+    //      webrtc://r.ossrs.net/live/livestream?schema=https
 
+    // or use plaintext, without SRTP:
 
+    //      webrtc://r.ossrs.net/live/livestream?encrypt=false
 
+    // or any other information, will pass-by in the query:
 
+    //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
 
+    //      webrtc://r.ossrs.net/live/livestream?token=xxx
 
+    self.play = async function(url) {
 
+        var conf = self.__internal.prepareUrl(url);
 
+        self.pc.addTransceiver("audio", {direction: "recvonly"});
 
+        self.pc.addTransceiver("video", {direction: "recvonly"});
 
+        //self.pc.addTransceiver("video", {direction: "recvonly"});
 
+        //self.pc.addTransceiver("audio", {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, tid: conf.tid, streamurl: conf.streamUrl,
 
+                clientip: null, sdp: offer.sdp
 
+            };
 
+            console.log("Generated offer: ", data);
 
+
 
+            const xhr = new XMLHttpRequest();
 
+            xhr.onload = function() {
 
+                if (xhr.readyState !== xhr.DONE) return;
 
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
 
+                const data = JSON.parse(xhr.responseText);
 
+                console.log("Got answer: ", data);
 
+                return data.code ? reject(xhr) : resolve(data);
 
+            }
 
+            xhr.open('POST', conf.apiUrl, true);
 
+            xhr.setRequestHeader('Content-type', 'application/json');
 
+            xhr.send(JSON.stringify(data));
 
+        });
 
+        await self.pc.setRemoteDescription(
 
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
 
+        );
 
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
 
+
 
+        return session;
 
+    };
 
+
 
+    // Close the player.
 
+    self.close = function() {
 
+        self.pc && self.pc.close();
 
+        self.pc = null;
 
+    };
 
+
 
+    // The callback when got remote track.
 
+    // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
 
+    self.ontrack = function (event) {
 
+        // https://webrtc.org/getting-started/remote-streams
 
+        self.stream.addTrack(event.track);
 
+    };
 
+
 
+    // 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 += '/';
 
+            }
 
+
 
+            var 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
 
+            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).slice(0, 7)
 
+            };
 
+        },
 
+        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.substring(1, a.pathname.lastIndexOf("/"));
 
+            var stream = a.pathname.slice(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.slice(app.indexOf("?"));
 
+                app = app.slice(0, app.indexOf("?"));
 
+
 
+                if (params.indexOf("vhost=") > 0) {
 
+                    vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
 
+                    if (vhost.indexOf("&") > 0) {
 
+                        vhost = vhost.slice(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.slice(0, url.indexOf("://"));
 
+            }
 
+
 
+            var port = a.port;
 
+            if (!port) {
 
+                // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
 
+                if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
 
+                    port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
 
+                }
 
+
 
+                // Guess by schema.
 
+                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);
 
+
 
+    // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
 
+    self.stream = new MediaStream();
 
+
 
+    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
 
+    self.pc.ontrack = function(event) {
 
+        if (self.ontrack) {
 
+            self.ontrack(event);
 
+        }
 
+    };
 
+
 
+    return self;
 
+}
 
+
 
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
 
+// Async-awat-prmise based SRS RTC Publisher by WHIP.
 
+function SrsRtcWhipWhepAsync() {
 
+    var self = {};
 
+
 
+    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
 
+    self.constraints = {
 
+        audio: true,
 
+        video: {
 
+            width: {ideal: 320, max: 576}
 
+        }
 
+    };
 
+
 
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
 
+    // @url The WebRTC url to publish with, for example:
 
+    //      http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
 
+    self.publish = async function (url) {
 
+        if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
 
+
 
+        self.pc.addTransceiver("audio", {direction: "sendonly"});
 
+        self.pc.addTransceiver("video", {direction: "sendonly"});
 
+
 
+        if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
 
+            throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
 
+        }
 
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
 
+
 
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
 
+        stream.getTracks().forEach(function (track) {
 
+            self.pc.addTrack(track);
 
+
 
+            // Notify about local track when stream is ok.
 
+            self.ontrack && self.ontrack({track: track});
 
+        });
 
+
 
+        var offer = await self.pc.createOffer();
 
+        await self.pc.setLocalDescription(offer);
 
+        const answer = await new Promise(function (resolve, reject) {
 
+            console.log("Generated offer: ", offer);
 
+
 
+            const xhr = new XMLHttpRequest();
 
+            xhr.onload = function() {
 
+                if (xhr.readyState !== xhr.DONE) return;
 
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
 
+                const data = xhr.responseText;
 
+                console.log("Got answer: ", data);
 
+                return data.code ? reject(xhr) : resolve(data);
 
+            }
 
+            xhr.open('POST', url, true);
 
+            xhr.setRequestHeader('Content-type', 'application/sdp');
 
+            xhr.send(offer.sdp);
 
+        });
 
+        await self.pc.setRemoteDescription(
 
+            new RTCSessionDescription({type: 'answer', sdp: answer})
 
+        );
 
+
 
+        return self.__internal.parseId(url, offer.sdp, answer);
 
+    };
 
+
 
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
 
+    // @url The WebRTC url to play with, for example:
 
+    //      http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
 
+    self.play = async function(url) {
 
+        if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
 
+
 
+        self.pc.addTransceiver("audio", {direction: "recvonly"});
 
+        self.pc.addTransceiver("video", {direction: "recvonly"});
 
+
 
+        var offer = await self.pc.createOffer();
 
+        await self.pc.setLocalDescription(offer);
 
+        const answer = await new Promise(function(resolve, reject) {
 
+            console.log("Generated offer: ", offer);
 
+
 
+            const xhr = new XMLHttpRequest();
 
+            xhr.onload = function() {
 
+                if (xhr.readyState !== xhr.DONE) return;
 
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
 
+                const data = xhr.responseText;
 
+                console.log("Got answer: ", data);
 
+                return data.code ? reject(xhr) : resolve(data);
 
+            }
 
+            xhr.open('POST', url, true);
 
+            xhr.setRequestHeader('Content-type', 'application/sdp');
 
+            xhr.send(offer.sdp);
 
+        });
 
+        await self.pc.setRemoteDescription(
 
+            new RTCSessionDescription({type: 'answer', sdp: answer})
 
+        );
 
+
 
+        return self.__internal.parseId(url, offer.sdp, answer);
 
+    };
 
+
 
+    // Close the publisher.
 
+    self.close = function () {
 
+        self.pc && self.pc.close();
 
+        self.pc = null;
 
+    };
 
+
 
+    // The callback when got local stream.
 
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
 
+    self.ontrack = function (event) {
 
+        // Add track to stream of SDK.
 
+        self.stream.addTrack(event.track);
 
+    };
 
+
 
+    self.pc = new RTCPeerConnection(null);
 
+
 
+    // To keep api consistent between player and publisher.
 
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
 
+    // @see https://webrtc.org/getting-started/media-devices
 
+    self.stream = new MediaStream();
 
+
 
+    // Internal APIs.
 
+    self.__internal = {
 
+        parseId: (url, offer, answer) => {
 
+            let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
 
+            sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
 
+            sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
 
+            sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
 
+
 
+            const a = document.createElement("a");
 
+            a.href = url;
 
+            return {
 
+                sessionid: sessionid, // Should be ice-ufrag of answer:offer.
 
+                simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
 
+            };
 
+        },
 
+    };
 
+
 
+    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
 
+    self.pc.ontrack = function(event) {
 
+        if (self.ontrack) {
 
+            self.ontrack(event);
 
+        }
 
+    };
 
+
 
+    return self;
 
+}
 
+
 
+// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
 
+// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
 
+function SrsRtcFormatSenders(senders, kind) {
 
+    var codecs = [];
 
+    senders.forEach(function (sender) {
 
+        var params = sender.getParameters();
 
+        params && params.codecs && params.codecs.forEach(function(c) {
 
+            if (kind && sender.track.kind !== kind) {
 
+                return;
 
+            }
 
+
 
+            if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
 
+                return;
 
+            }
 
+
 
+            var s = '';
 
+
 
+            s += c.mimeType.replace('audio/', '').replace('video/', '');
 
+            s += ', ' + c.clockRate + 'HZ';
 
+            if (sender.track.kind === "audio") {
 
+                s += ', channels: ' + c.channels;
 
+            }
 
+            s += ', pt: ' + c.payloadType;
 
+
 
+            codecs.push(s);
 
+        });
 
+    });
 
+    return codecs.join(", ");
 
+}
 
+

+ 2 - 2
src/views/camera/index.vue
View File 

@@ -3,11 +3,11 @@
 
     <!--安防摄像头接入-->
 
     <!--安防摄像头接入-->
 
     <el-col :span="20" :xs="15">
 
     <el-col :span="20" :xs="15">
 
       <ContentWrap>
 
       <ContentWrap>
 
-        <el-input v-model="rtspAddr" class="!w-450px" placeholder="请输入RTSP/RTMP视频流服务地址" />
 
+        <el-input v-model="rtspAddr" class="!w-450px" placeholder="请输入RTSP/RTMP视频拉流地址" />
 
         <el-button @click="handleChange">接入</el-button>
 
         <el-button @click="handleChange">接入</el-button>
 
       </ContentWrap>
 
       </ContentWrap>
 
       <ContentWrap>
 
       <ContentWrap>
 
-        <video id="video" autoplay width="500" height="400"></video>
 
+        <video id="video" autoplay controls width="500" height="400"></video>
 
       </ContentWrap>
 
       </ContentWrap>
 
     </el-col>
 
     </el-col>
 
   </el-row>
 
   </el-row>

+ 66 - 0
src/views/camera/push.vue
View File 

@@ -0,0 +1,66 @@
 
+<template>
 
+  <el-row :gutter="15">
 
+    <!--安防摄像头接入-->
 
+    <el-col :span="20" :xs="15">
 
+      <ContentWrap>
 
+        <el-input v-model="url" class="!w-450px" placeholder="请输入视频流ID" />
 
+        <el-button @click="startPlay">接入</el-button>
 
+      </ContentWrap>
 
+      <ContentWrap>
 
+        <video
 
+          ref="videoPlayer"
 
+          controls="true" autoplay="true"
 
+          width="500"
 
+          height="400"
 
+        ></video>
 
+      </ContentWrap>
 
+    </el-col>
 
+  </el-row>
 
+</template>
 
+
 
+<script lang="ts" setup>
 
+import { onMounted, onUnmounted, ref } from 'vue'
 
+let sdk = null
 
+const videoPlayer = ref(null)
 
+const baseUrl = 'http://123.60.223.250:1985/rtc/v1/whep/?app=live&stream='
 
+let url = `caoyang001`
 
+
 
+onMounted(() => {
 
+  startPlay()
 
+})
 
+
 
+onUnmounted(() => {
 
+  if (sdk) {
 
+    sdk.close()
 
+  }
 
+  sdk = null
 
+})
 
+
 
+const startPlay = async () => {
 
+  try {
 
+    console.log(url)
 
+    if (url) {
 
+      if (sdk) {
 
+        sdk.close()
 
+        sdk = null
 
+      }
 
+      sdk = new SrsRtcWhipWhepAsync()
 
+      sdk.play(baseUrl+url)
 
+        .then(function (session) {
 
+          videoPlayer.value.srcObject = sdk.stream
 
+        })
 
+        .catch(function (reason) {
 
+          console.error(reason)
 
+        })
 
+    } else {
 
+      ElMessage.error(
 
+        '请填写正确地址，地址格式为：http://123.60.223.250:1985/rtc/v1/whep/?app=live&stream=ID'
 
+      )
 
+    }
 
+  } catch (error) {
 
+    console.log(error)
 
+  }
 
+}
 
+</script>
 
+
 
+<style scoped></style>