a3a2fa5ceb
**Problem**: Newly created sources (RTMP/SRT/RTC/RTSP) were being immediately marked as "dead" and deleted by the cleanup timer before publishers could connect, causing "new live source, dead=1" errors. **Root Cause**: All source constructors initialized `stream_die_at_ = 0`, causing `stream_is_dead()` to return `true` immediately since current time was always greater than `0 + 3 seconds`. **Solution**: Changed all four source constructors to initialize `stream_die_at_ = srs_time_now_cached()`, giving newly created sources a proper 3-second grace period before cleanup.
336 lines
13 KiB
JavaScript
336 lines
13 KiB
JavaScript
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//
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// Copyright (c) 2013-2025 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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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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// 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: 720},
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//width: {ideal: 720, max: 1080},
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//width: 1280, height: 720, frameRate: 30,
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}
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};
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// Store media streams to stop tracks when closing.
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self.displayStream = null;
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self.userStream = null;
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// Store the WHIP session resource URL from Location header for cleanup.
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self.resourceUrl = null;
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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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// @options The options to control playing, supports:
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// camera: boolean, whether capture video from camera, default to true.
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// screen: boolean, whether capture video from screen, default to false.
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// audio: boolean, whether play audio, default to true.
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// vcodec: string, video codec to use (e.g., 'h264', 'vp9', 'av1'), default to undefined.
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// acodec: string, audio codec to use (e.g., 'opus', 'pcmu', 'pcma'), default to undefined.
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self.publish = async function (url, options) {
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if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
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const hasAudio = options?.audio ?? true;
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const useCamera = options?.camera ?? true;
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const useScreen = options?.screen ?? false;
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const vcodec = options?.vcodec;
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const acodec = options?.acodec;
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if (!hasAudio && !useCamera && !useScreen) throw new Error(`The camera, screen and audio can't be false at the same time`);
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if (hasAudio) {
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self.pc.addTransceiver("audio", {direction: "sendonly"});
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} else {
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self.constraints.audio = false;
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}
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if (useCamera || useScreen) {
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self.pc.addTransceiver("video", {direction: "sendonly"});
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}
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if (!useCamera) {
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self.constraints.video = false;
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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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if (useScreen) {
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self.displayStream = await navigator.mediaDevices.getDisplayMedia({
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video: true
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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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self.displayStream.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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}
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if (useCamera || hasAudio) {
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self.userStream = await navigator.mediaDevices.getUserMedia(self.constraints);
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self.userStream.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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}
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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// Filter codecs if specified
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if (vcodec || acodec) {
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offer.sdp = self.__internal.filterCodec(offer.sdp, vcodec, acodec);
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console.log(`Filtered codecs (vcodec=${vcodec}, acodec=${acodec}): ${offer.sdp}`);
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}
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const answer = await new Promise(function (resolve, reject) {
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console.log(`Generated offer: ${offer.sdp}`);
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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 = xhr.responseText;
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console.log("Got answer: ", data);
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// Extract Location header for WHIP session resource URL.
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const location = xhr.getResponseHeader('Location');
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if (location) {
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self.resourceUrl = new URL(location, url).href;
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console.log(`WHIP session resource URL: ${self.resourceUrl}`);
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}
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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/whep/?app=live&stream=livestream
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// @options The options to control playing, supports:
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// videoOnly: boolean, whether only play video, default to false.
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// audioOnly: boolean, whether only play audio, default to false.
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// vcodec: string, video codec to use (e.g., 'h264', 'vp9', 'av1'), default to undefined.
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// acodec: string, audio codec to use (e.g., 'opus', 'pcmu', 'pcma'), default to undefined.
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self.play = async function(url, options) {
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if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
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if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
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const vcodec = options?.vcodec;
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const acodec = options?.acodec;
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if (!options?.videoOnly) self.pc.addTransceiver("audio", {direction: "recvonly"});
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if (!options?.audioOnly) self.pc.addTransceiver("video", {direction: "recvonly"});
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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// Filter codecs if specified
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if (vcodec || acodec) {
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offer.sdp = self.__internal.filterCodec(offer.sdp, vcodec, acodec);
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console.log(`Filtered codecs (vcodec=${vcodec}, acodec=${acodec}): ${offer.sdp}`);
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}
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const answer = await new Promise(function(resolve, reject) {
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console.log(`Generated offer: ${offer.sdp}`);
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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 = xhr.responseText;
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console.log("Got answer: ", data);
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// Extract Location header for WHEP session resource URL.
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const location = xhr.getResponseHeader('Location');
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if (location) {
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self.resourceUrl = new URL(location, url).href;
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console.log(`WHEP session resource URL: ${self.resourceUrl}`);
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}
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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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// Stop all media tracks to release camera/microphone.
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if (self.displayStream) {
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self.displayStream.getTracks().forEach(function (track) {
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track.stop();
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});
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self.displayStream = null;
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}
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if (self.userStream) {
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self.userStream.getTracks().forEach(function (track) {
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track.stop();
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});
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self.userStream = null;
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}
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// Send DELETE request to WHIP session resource URL to cleanup server resources.
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if (self.resourceUrl) {
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const xhr = new XMLHttpRequest();
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xhr.open('DELETE', self.resourceUrl, true);
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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) {
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console.log(`WHIP session deleted: ${self.resourceUrl}`);
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} else {
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console.warn(`Failed to delete WHIP session: ${self.resourceUrl}, status: ${xhr.status}`);
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}
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};
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xhr.onerror = function() {
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console.warn(`Error deleting WHIP session: ${self.resourceUrl}`);
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};
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xhr.send();
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self.resourceUrl = 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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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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filterCodec: (sdp, vcodec, acodec) => {
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// Filter video codec if specified
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if (vcodec) {
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const vcodecUpper = vcodec.toUpperCase();
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sdp = sdp.split('\n').filter(line => {
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// Keep all non-video lines
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if (!line.startsWith('a=rtpmap:') && !line.startsWith('a=rtcp-fb:') &&
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!line.startsWith('a=fmtp:')) {
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return true;
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}
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// For video codec lines, only keep the specified codec
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if (line.includes('video/')) {
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return line.toUpperCase().includes(vcodecUpper);
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}
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return true;
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}).join('\n');
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}
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// Filter audio codec if specified
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if (acodec) {
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const acodecUpper = acodec.toUpperCase();
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sdp = sdp.split('\n').filter(line => {
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// Keep all non-audio lines
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if (!line.startsWith('a=rtpmap:') && !line.startsWith('a=rtcp-fb:') &&
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!line.startsWith('a=fmtp:')) {
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return true;
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}
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// For audio codec lines, only keep the specified codec
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if (line.includes('audio/')) {
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return line.toUpperCase().includes(acodecUpper);
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}
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return true;
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}).join('\n');
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}
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return sdp;
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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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// https://developer.mozilla.org/en-US/docs/Web/API/RTCStatsReport
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function SrsRtcFormatStats(stats, kind) {
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var codecs = [];
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stats.forEach((report) => {
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if (report.type === 'codec' && report.mimeType?.toLowerCase().startsWith(kind)) {
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var s = '';
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s += report.mimeType.split('/')[1] || report.mimeType;
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if (report.clockRate) {
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s += ', ' + report.clockRate + 'HZ';
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}
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if (kind === 'audio' && report.channels) {
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s += ', channels: ' + report.channels;
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}
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if (report.payloadType) {
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s += ', pt: ' + report.payloadType;
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}
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codecs.push(s);
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}
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});
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return codecs.join(", ");
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} |