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@@ -0,0 +1,681 @@
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+
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+//
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+// Copyright (c) 2013-2021 Winlin
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+//
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+// SPDX-License-Identifier: MIT
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+//
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+
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+'use strict';
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+
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+function SrsError(name, message) {
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+ this.name = name;
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+ this.message = message;
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+ this.stack = (new Error()).stack;
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+}
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+SrsError.prototype = Object.create(Error.prototype);
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+SrsError.prototype.constructor = SrsError;
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+
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+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
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+// Async-awat-prmise based SRS RTC Publisher.
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+function SrsRtcPublisherAsync() {
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+ var self = {};
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+
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+ // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
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+ self.constraints = {
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+ audio: true,
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+ video: {
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+ width: {ideal: 320, max: 576}
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+ }
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+ };
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+
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+ // @see https://github.com/rtcdn/rtcdn-draft
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+ // @url The WebRTC url to play with, for example:
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+ // webrtc://r.ossrs.net/live/livestream
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+ // or specifies the API port:
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+ // webrtc://r.ossrs.net:11985/live/livestream
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+ // or autostart the publish:
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+ // webrtc://r.ossrs.net/live/livestream?autostart=true
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+ // or change the app from live to myapp:
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+ // webrtc://r.ossrs.net:11985/myapp/livestream
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+ // or change the stream from livestream to mystream:
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+ // webrtc://r.ossrs.net:11985/live/mystream
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+ // or set the api server to myapi.domain.com:
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+ // webrtc://myapi.domain.com/live/livestream
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+ // or set the candidate(eip) of answer:
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+ // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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+ // or force to access https API:
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+ // webrtc://r.ossrs.net/live/livestream?schema=https
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+ // or use plaintext, without SRTP:
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+ // webrtc://r.ossrs.net/live/livestream?encrypt=false
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+ // or any other information, will pass-by in the query:
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+ // webrtc://r.ossrs.net/live/livestream?vhost=xxx
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+ // webrtc://r.ossrs.net/live/livestream?token=xxx
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+ self.publish = async function (url) {
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+ var conf = self.__internal.prepareUrl(url);
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+ self.pc.addTransceiver("audio", {direction: "sendonly"});
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+ self.pc.addTransceiver("video", {direction: "sendonly"});
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+ //self.pc.addTransceiver("video", {direction: "sendonly"});
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+ //self.pc.addTransceiver("audio", {direction: "sendonly"});
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+
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+ if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
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+ throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
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+ }
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+ var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
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+
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+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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+ stream.getTracks().forEach(function (track) {
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+ self.pc.addTrack(track);
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+
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+ // Notify about local track when stream is ok.
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+ self.ontrack && self.ontrack({track: track});
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+ });
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+
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+ var offer = await self.pc.createOffer();
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+ await self.pc.setLocalDescription(offer);
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+ var session = await new Promise(function (resolve, reject) {
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+ // @see https://github.com/rtcdn/rtcdn-draft
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+ var data = {
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+ api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
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+ clientip: null, sdp: offer.sdp
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+ };
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+ console.log("Generated offer: ", data);
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+
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+ const xhr = new XMLHttpRequest();
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+ xhr.onload = function() {
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+ if (xhr.readyState !== xhr.DONE) return;
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+ if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
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+ const data = JSON.parse(xhr.responseText);
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+ console.log("Got answer: ", data);
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+ return data.code ? reject(xhr) : resolve(data);
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+ }
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+ xhr.open('POST', conf.apiUrl, true);
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+ xhr.setRequestHeader('Content-type', 'application/json');
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+ xhr.send(JSON.stringify(data));
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+ });
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+ await self.pc.setRemoteDescription(
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+ new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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+ );
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+ session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
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+
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+ return session;
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+ };
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+
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+ // Close the publisher.
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+ self.close = function () {
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+ self.pc && self.pc.close();
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+ self.pc = null;
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+ };
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+
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+ // The callback when got local stream.
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+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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+ self.ontrack = function (event) {
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+ // Add track to stream of SDK.
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+ self.stream.addTrack(event.track);
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+ };
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+
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+ // Internal APIs.
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+ self.__internal = {
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+ defaultPath: '/rtc/v1/publish/',
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+ prepareUrl: function (webrtcUrl) {
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+ var urlObject = self.__internal.parse(webrtcUrl);
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+
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+ // If user specifies the schema, use it as API schema.
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+ var schema = urlObject.user_query.schema;
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+ schema = schema ? schema + ':' : window.location.protocol;
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+
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+ var port = urlObject.port || 1985;
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+ if (schema === 'https:') {
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+ port = urlObject.port || 443;
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+ }
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+
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+ // @see https://github.com/rtcdn/rtcdn-draft
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+ var api = urlObject.user_query.play || self.__internal.defaultPath;
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+ if (api.lastIndexOf('/') !== api.length - 1) {
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+ api += '/';
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+ }
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+
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+ var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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+ for (var key in urlObject.user_query) {
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+ if (key !== 'api' && key !== 'play') {
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+ apiUrl += '&' + key + '=' + urlObject.user_query[key];
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+ }
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+ }
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+ // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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+ apiUrl = apiUrl.replace(api + '&', api + '?');
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+
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+ var streamUrl = urlObject.url;
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+
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+ return {
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+ apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
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+ tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
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+ };
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+ },
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+ parse: function (url) {
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+ // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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+ var a = document.createElement("a");
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+ a.href = url.replace("rtmp://", "http://")
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+ .replace("webrtc://", "http://")
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+ .replace("rtc://", "http://");
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+
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+ var vhost = a.hostname;
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+ var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
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+ var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
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+
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+ // parse the vhost in the params of app, that srs supports.
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+ app = app.replace("...vhost...", "?vhost=");
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+ if (app.indexOf("?") >= 0) {
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+ var params = app.slice(app.indexOf("?"));
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+ app = app.slice(0, app.indexOf("?"));
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+
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+ if (params.indexOf("vhost=") > 0) {
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+ vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
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+ if (vhost.indexOf("&") > 0) {
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+ vhost = vhost.slice(0, vhost.indexOf("&"));
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+ }
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+ }
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+ }
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+
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+ // when vhost equals to server, and server is ip,
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+ // the vhost is __defaultVhost__
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+ if (a.hostname === vhost) {
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+ var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
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+ if (re.test(a.hostname)) {
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+ vhost = "__defaultVhost__";
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+ }
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+ }
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+
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+ // parse the schema
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+ var schema = "rtmp";
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+ if (url.indexOf("://") > 0) {
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+ schema = url.slice(0, url.indexOf("://"));
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+ }
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+
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+ var port = a.port;
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+ if (!port) {
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+ // Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
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+ if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
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+ port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
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+ }
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+
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+ // Guess by schema.
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+ if (schema === 'http') {
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+ port = 80;
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+ } else if (schema === 'https') {
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+ port = 443;
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+ } else if (schema === 'rtmp') {
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+ port = 1935;
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+ }
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+ }
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+
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+ var ret = {
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+ url: url,
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+ schema: schema,
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+ server: a.hostname, port: port,
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+ vhost: vhost, app: app, stream: stream
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+ };
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+ self.__internal.fill_query(a.search, ret);
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+
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+ // For webrtc API, we use 443 if page is https, or schema specified it.
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+ if (!ret.port) {
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+ if (schema === 'webrtc' || schema === 'rtc') {
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+ if (ret.user_query.schema === 'https') {
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+ ret.port = 443;
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+ } else if (window.location.href.indexOf('https://') === 0) {
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+ ret.port = 443;
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+ } else {
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+ // For WebRTC, SRS use 1985 as default API port.
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+ ret.port = 1985;
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+ }
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+ }
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+ }
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+
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+ return ret;
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+ },
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+ fill_query: function (query_string, obj) {
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+ // pure user query object.
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+ obj.user_query = {};
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+
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+ if (query_string.length === 0) {
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+ return;
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+ }
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+
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+ // split again for angularjs.
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+ if (query_string.indexOf("?") >= 0) {
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+ query_string = query_string.split("?")[1];
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+ }
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+
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+ var queries = query_string.split("&");
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+ for (var i = 0; i < queries.length; i++) {
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+ var elem = queries[i];
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+
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+ var query = elem.split("=");
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+ obj[query[0]] = query[1];
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+ obj.user_query[query[0]] = query[1];
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+ }
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+
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+ // alias domain for vhost.
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+ if (obj.domain) {
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+ obj.vhost = obj.domain;
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+ }
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+ }
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+ };
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+
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+ self.pc = new RTCPeerConnection(null);
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+
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+ // To keep api consistent between player and publisher.
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+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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+ // @see https://webrtc.org/getting-started/media-devices
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+ self.stream = new MediaStream();
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+
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+ return self;
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+}
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+
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+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
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+// Async-await-promise based SRS RTC Player.
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+function SrsRtcPlayerAsync() {
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+ var self = {};
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+
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+ // @see https://github.com/rtcdn/rtcdn-draft
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+ // @url The WebRTC url to play with, for example:
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+ // webrtc://r.ossrs.net/live/livestream
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+ // or specifies the API port:
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+ // webrtc://r.ossrs.net:11985/live/livestream
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+ // webrtc://r.ossrs.net:80/live/livestream
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+ // or autostart the play:
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+ // webrtc://r.ossrs.net/live/livestream?autostart=true
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+ // or change the app from live to myapp:
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+ // webrtc://r.ossrs.net:11985/myapp/livestream
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+ // or change the stream from livestream to mystream:
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+ // webrtc://r.ossrs.net:11985/live/mystream
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+ // or set the api server to myapi.domain.com:
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+ // webrtc://myapi.domain.com/live/livestream
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+ // or set the candidate(eip) of answer:
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+ // webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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+ // or force to access https API:
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+ // webrtc://r.ossrs.net/live/livestream?schema=https
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+ // or use plaintext, without SRTP:
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+ // webrtc://r.ossrs.net/live/livestream?encrypt=false
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+ // or any other information, will pass-by in the query:
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+ // webrtc://r.ossrs.net/live/livestream?vhost=xxx
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+ // webrtc://r.ossrs.net/live/livestream?token=xxx
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+ self.play = async function(url) {
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+ var conf = self.__internal.prepareUrl(url);
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+ self.pc.addTransceiver("audio", {direction: "recvonly"});
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+ self.pc.addTransceiver("video", {direction: "recvonly"});
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+ //self.pc.addTransceiver("video", {direction: "recvonly"});
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+ //self.pc.addTransceiver("audio", {direction: "recvonly"});
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+
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+ var offer = await self.pc.createOffer();
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+ await self.pc.setLocalDescription(offer);
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+ var session = await new Promise(function(resolve, reject) {
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+ // @see https://github.com/rtcdn/rtcdn-draft
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+ var data = {
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+ api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
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+ clientip: null, sdp: offer.sdp
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+ };
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+ console.log("Generated offer: ", data);
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+
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+ const xhr = new XMLHttpRequest();
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+ xhr.onload = function() {
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+ if (xhr.readyState !== xhr.DONE) return;
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+ if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
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+ const data = JSON.parse(xhr.responseText);
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+ console.log("Got answer: ", data);
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+ return data.code ? reject(xhr) : resolve(data);
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+ }
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+ xhr.open('POST', conf.apiUrl, true);
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+ xhr.setRequestHeader('Content-type', 'application/json');
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+ xhr.send(JSON.stringify(data));
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+ });
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+ await self.pc.setRemoteDescription(
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+ new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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+ );
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+ session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
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+
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+ return session;
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+ };
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+
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+ // Close the player.
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+ self.close = function() {
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+ self.pc && self.pc.close();
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+ self.pc = null;
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+ };
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+
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+ // The callback when got remote track.
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+ // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
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+ self.ontrack = function (event) {
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+ // https://webrtc.org/getting-started/remote-streams
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+ self.stream.addTrack(event.track);
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+ };
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+
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+ // Internal APIs.
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+ self.__internal = {
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+ defaultPath: '/rtc/v1/play/',
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+ prepareUrl: function (webrtcUrl) {
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+ var urlObject = self.__internal.parse(webrtcUrl);
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+
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+ // If user specifies the schema, use it as API schema.
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+ var schema = urlObject.user_query.schema;
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+ schema = schema ? schema + ':' : window.location.protocol;
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+
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+ var port = urlObject.port || 1985;
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+ if (schema === 'https:') {
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+ port = urlObject.port || 443;
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+ }
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+
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+ // @see https://github.com/rtcdn/rtcdn-draft
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+ var api = urlObject.user_query.play || self.__internal.defaultPath;
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+ if (api.lastIndexOf('/') !== api.length - 1) {
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+ api += '/';
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+ }
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+
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+ 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(", ");
|
|
|
+}
|
|
|
+
|
