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Support WHIP and WHEP player. v5.0.147 and v6.0.35 (#3460)
Co-authored-by: chundonglinlin <chundonglinlin@163.com> Co-authored-by: panda <542638787@qq.com>
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Winlin
GitHub
parent
5067e220ca
commit
c001acaae9
12 changed files with 456 additions and 10 deletions
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@ -134,14 +134,14 @@ function SrsRtcPublisherAsync() {
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api += '/';
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}
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apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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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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var apiUrl = apiUrl.replace(api + '&', api + '?');
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apiUrl = apiUrl.replace(api + '&', api + '?');
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var streamUrl = urlObject.url;
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@ -369,14 +369,14 @@ function SrsRtcPlayerAsync() {
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api += '/';
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}
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apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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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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var apiUrl = apiUrl.replace(api + '&', api + '?');
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apiUrl = apiUrl.replace(api + '&', api + '?');
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var streamUrl = urlObject.url;
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@ -510,6 +510,145 @@ function SrsRtcPlayerAsync() {
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return self;
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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 by WHIP.
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function SrsRtcWhipWhepAsync() {
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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://datatracker.ietf.org/doc/draft-ietf-wish-whip/
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// @url The WebRTC url to publish with, for example:
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// http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
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self.publish = async function (url) {
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if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${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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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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// @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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const answer = await new Promise(function (resolve, reject) {
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console.log("Generated offer: ", offer);
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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) return reject(xhr);
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const data = 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', url, true);
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xhr.setRequestHeader('Content-type', 'application/sdp');
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xhr.send(offer.sdp);
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: answer})
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);
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return self.__internal.parseId(url, offer.sdp, answer);
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};
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// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
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// @url The WebRTC url to play with, for example:
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// http://localhost:1985/rtc/v1/whip-play/?app=live&stream=livestream
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self.play = async function(url) {
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if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${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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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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const answer = await new Promise(function(resolve, reject) {
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console.log("Generated offer: ", offer);
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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) return reject(xhr);
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const data = 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', url, true);
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xhr.setRequestHeader('Content-type', 'application/sdp');
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xhr.send(offer.sdp);
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: answer})
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);
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return self.__internal.parseId(url, offer.sdp, answer);
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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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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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// Internal APIs.
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self.__internal = {
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parseId: (url, offer, answer) => {
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let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
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sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
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sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
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sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
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const a = document.createElement("a");
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a.href = url;
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return {
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sessionid: sessionid, // Should be ice-ufrag of answer:offer.
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simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
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};
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},
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};
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// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
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self.pc.ontrack = function(event) {
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if (self.ontrack) {
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self.ontrack(event);
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}
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};
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return self;
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}
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// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
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// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
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function SrsRtcFormatSenders(senders, kind) {
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