easy-player-pro 0.1.5 → 0.1.7

package/lib/build/srs.sdk.js

@@ -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(", ");
+}
+