superxxd
2 years ago
7 changed files with 635 additions and 35 deletions
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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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'use strict'; |
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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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// 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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// @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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var stream = await navigator.mediaDevices.getUserMedia(self.constraints); |
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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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// 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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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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$.ajax({ |
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type: "POST", url: conf.apiUrl, data: JSON.stringify(data), |
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contentType: 'application/json', dataType: 'json' |
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}).done(function (data) { |
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console.log("Got answer: ", data); |
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if (data.code) { |
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reject(data); |
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return; |
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} |
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resolve(data); |
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}).fail(function (reason) { |
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reject(reason); |
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}); |
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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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return session; |
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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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// 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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// 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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// 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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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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// @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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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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var apiUrl = apiUrl.replace(api + '&', api + '?'); |
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var streamUrl = urlObject.url; |
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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).substr(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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var vhost = a.hostname; |
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var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1); |
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var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1); |
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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.substr(app.indexOf("?")); |
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app = app.substr(0, app.indexOf("?")); |
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if (params.indexOf("vhost=") > 0) { |
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vhost = params.substr(params.indexOf("vhost=") + "vhost=".length); |
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if (vhost.indexOf("&") > 0) { |
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vhost = vhost.substr(0, vhost.indexOf("&")); |
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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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// parse the schema
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var schema = "rtmp"; |
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if (url.indexOf("://") > 0) { |
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schema = url.substr(0, url.indexOf("://")); |
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} |
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var port = a.port; |
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if (!port) { |
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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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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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// 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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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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if (query_string.length === 0) { |
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return; |
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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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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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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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// 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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self.pc = new RTCPeerConnection(null); |
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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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return self; |
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} |
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var datachannel=null; |
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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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const config = { |
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bundlePolicy: 'balanced', |
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// certificates?: RTCCertificate[];
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// iceCandidatePoolSize?: number;
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iceTransportPolicy: "relay",// all
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rtcpMuxPolicy : 'negotiate', |
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iceServers: [ |
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{ |
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urls: "turn:192.168.1.102:3478", |
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username: "metartc", |
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credential: "metartc" |
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} |
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] |
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}; |
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self.pc = new RTCPeerConnection(config); |
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* */ |
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self.pc = new RTCPeerConnection(null); |
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self.pc.onconnectionstatechange=function(event){ |
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console.log("connection state change: ", self.pc.connectionState); |
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} |
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self.pc.onicecandidate = async (ev) => { |
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console.log('=======>' + JSON.stringify(ev.candidate)); |
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}; |
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datachannel=self.pc.createDataChannel('chat'); |
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datachannel.onopen = function(event) { |
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console.log("datachannel onopen: ", event.data); |
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} |
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datachannel.onmessage = function(event) { |
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console.log("receive message: ", event.data); |
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$('#datachannel_recv').val(event.data); |
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} |
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datachannel.onerror=function(event) { |
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console.log("datachannel error: ", event.data); |
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} |
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datachannel.onclose=function(event) { |
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console.log("datachannel close: "); |
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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 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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console.log("conf.apiUrl: ", conf.apiUrl); |
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self.pc.addTransceiver("audio", {direction: "recvonly"}); |
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self.pc.addTransceiver("video", {direction: "recvonly"}); |
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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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//text/plain application/json
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$.ajax({ |
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type: "POST", url: conf.apiUrl, data: offer.sdp+"}", |
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contentType:'text/plain', dataType: 'json', |
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crossDomain:true |
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}).done(function(data) { |
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if (data.code) { |
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reject(data); return; |
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} |
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console.log("Got sdp: ", data.sdp); |
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resolve(data); |
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}).fail(function(reason){ |
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reject(reason); |
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}); |
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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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return session; |
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}; |
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// Close the player.
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self.close = function() { |
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if(datachannel) { |
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datachannel.close(); |
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datachannel=null; |
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} |
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self.pc && self.pc.close(); |
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self.pc = null; |
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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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// 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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var schema="http:"; |
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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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// @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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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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var apiUrl = apiUrl.replace(api + '&', api + '?'); |
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var streamUrl = urlObject.url; |
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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).substr(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.substr(1, a.pathname.lastIndexOf("/") - 1); |
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var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1); |
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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.substr(app.indexOf("?")); |
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app = app.substr(0, app.indexOf("?")); |
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|
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if (params.indexOf("vhost=") > 0) { |
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vhost = params.substr(params.indexOf("vhost=") + "vhost=".length); |
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if (vhost.indexOf("&") > 0) { |
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vhost = vhost.substr(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.substr(0, url.indexOf("://")); |
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} |
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var port = a.port; |
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if (!port) { |
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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) { |
|||
// 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; |
|||
} |
|||
} |
|||
}; |
|||
|
|||
|
|||
|
|||
// 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; |
|||
} |
|||
|
|||
// 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(", "); |
|||
} |
|||
|
|||
Loading…
Reference in new issue