SquashSRS4: Add one to one RTC demo.

trunk/3rdparty/signaling/www/demos/js/srs.sdk.js

@@ -0,0 +1,509 @@
+
+/**
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2013-2021 Winlin
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
+ * the Software, and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
+ * FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
+ * COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
+ * IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+ * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+'use strict';
+
+// 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 = {};
+
+    // @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(ip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?eip=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"});
+
+        var stream = await navigator.mediaDevices.getUserMedia(
+            {audio: true, video: {height: {max: 320}}}
+        );
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+        stream.getTracks().forEach(function (track) {
+            self.pc.addTrack(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);
+
+            $.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);
+            });
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+        );
+        session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
+
+        // Notify about local stream when success.
+        self.onaddstream && self.onaddstream({stream: stream});
+
+        return session;
+    };
+
+    // Close the publisher.
+    self.close = function () {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got local stream.
+    self.onaddstream = function (event) {
+    };
+
+    // 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 += '/';
+            }
+
+            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: new Date().getTime().toString(16)
+            };
+        },
+        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);
+
+    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
+    // 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(ip) of answer:
+    //      webrtc://r.ossrs.net/live/livestream?eip=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"});
+
+        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);
+
+            $.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);
+            });
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: session.sdp})
+        );
+        return session;
+    };
+
+    // Close the player.
+    self.close = function() {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // 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,
+                tid: new Date().getTime().toString(16)
+            };
+        },
+        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;
+}
+
+// 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) {
+        sender.getParameters().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(", ");
+}
+

trunk/3rdparty/signaling/www/demos/js/srs.sig.js

@@ -0,0 +1,120 @@
+
+/**
+ * The MIT License (MIT)
+ *
+ * Copyright (c) 2013-2021 Winlin
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
+ * the Software, and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
+ * FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
+ * COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
+ * IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+ * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+'use strict';
+
+// Async-await-promise based SRS RTC Signaling.
+function SrsRtcSignalingAsync() {
+    var self = {};
+
+    // The schema is ws or wss, host is ip or ip:port, display is nickname
+    // of user to join the room.
+    self.connect = async function (schema, host, room, display) {
+        var url = schema + '://' + host + '/sig/v1/rtc';
+        self.ws = new WebSocket(url + '?room=' + room + '&display=' + display);
+
+        self.ws.onmessage = function(event) {
+            var r = JSON.parse(event.data);
+            var promise = self._internals.msgs[r.tid];
+            if (promise) {
+                promise.resolve(r.msg);
+                delete self._internals.msgs[r.tid];
+            } else {
+                self.onmessage(r.msg);
+            }
+        };
+
+        return new Promise(function (resolve, reject) {
+            self.ws.onopen = function (event) {
+                resolve(event);
+            };
+
+            self.ws.onerror = function (event) {
+                reject(event);
+            };
+        });
+    };
+
+    // The message is a json object.
+    self.send = async function (message) {
+        return new Promise(function (resolve, reject) {
+            var r = {tid: new Date().getTime().toString(16), msg: message};
+            self._internals.msgs[r.tid] = {resolve: resolve, reject: reject};
+            self.ws.send(JSON.stringify(r));
+        });
+    };
+
+    self.close = function () {
+        self.ws && self.ws.close();
+        self.ws = null;
+
+        for (const tid in self._internals.msgs) {
+            var promise = self._internals.msgs[tid];
+            promise.reject('close');
+        }
+    };
+
+    // The callback when got messages from signaling server.
+    self.onmessage = function (msg) {
+    };
+
+    self._internals = {
+        // Key is tid, value is object {resolve, reject, response}.
+        msgs: {}
+    };
+
+    return self;
+}
+
+// Parse params in query string.
+function SrsRtcSignalingParse(location) {
+    let query = location.href.split('?')[1];
+    query = query? '?' + query : null;
+
+    let wsSchema = location.href.split('wss=')[1];
+    wsSchema = wsSchema? wsSchema.split('&')[0] : (location.protocol === 'http:'? 'ws' : 'wss');
+
+    let wsHost = location.href.split('wsh=')[1];
+    wsHost = wsHost? wsHost.split('&')[0] : location.hostname;
+
+    let wsPort = location.href.split('wsp=')[1];
+    wsPort = wsPort? wsPort.split('&')[0] : location.host.split(':')[1];
+    wsHost = wsPort? wsHost.split(':')[0] + ':' + wsPort : wsHost;
+
+    let host = location.href.split('host=')[1];
+    host = host? host.split('&')[0] : location.hostname;
+
+    let room = location.href.split('room=')[1];
+
+    let display = location.href.split('display=')[1];
+    display = display? display.split('&')[0] : new Date().getTime().toString(16).substr(3);
+
+    let autostart = location.href.split('autostart=')[1];
+    autostart = autostart && autostart.split('&')[0] === 'true';
+
+    return {
+        query: query, wsSchema: wsSchema, wsHost: wsHost, host: host,
+        room: room, display: display, autostart: autostart,
+    };
+}