diff --git a/.gitignore b/.gitignore index 50715e164..f7e6541ff 100644 --- a/.gitignore +++ b/.gitignore @@ -34,3 +34,5 @@ .idea .DS_Store +/cmake-build-debug/ +/CMakeLists.txt diff --git a/AUTHORS.txt b/AUTHORS.txt index 099d3d620..81290103e 100644 --- a/AUTHORS.txt +++ b/AUTHORS.txt @@ -84,3 +84,5 @@ CONTRIBUTORS ordered by first contribution. * xbpeng121<53243357+xbpeng121@users.noreply.github.com> * johzzy * stone +* cfw11<34058899+cfw11@users.noreply.github.com> +* louis.xia<68469352@qq.com> diff --git a/README.md b/README.md index 96d7b1921..56c7e8600 100755 --- a/README.md +++ b/README.md @@ -9,9 +9,10 @@ SRS/4.0，[Leo][release4]，是一个简单高效的实时视频服务器，支 SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181. -> Note: SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing]. +SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing]. + ## Usage Run SRS by [docker][docker-srs4], images is [here](https://hub.docker.com/r/ossrs/srs/tags) or [there](https://cr.console.aliyun.com/repository/cn-hangzhou/ossrs/srs/images), @@ -23,6 +24,7 @@ docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \ ossrs/srs:v4.0.117 ./objs/srs -c conf/srs.conf ``` + Or build SRS from source(or [mirrors](#mirrors)), by CentOS7(or Linux([CN][v4_CN_Build],[EN][v4_EN_Build])): ``` @@ -77,15 +79,20 @@ Other important wiki: ## Ports -The ports used by SRS: +The ports used by SRS, kernel services: + +* tcp://1935, for RTMP live streaming server([CN][v4_CN_DeliveryRTMP],[EN][v4_EN_DeliveryRTMP]). +* tcp://1985, HTTP API server, for HTTP-API([CN][v4_CN_HTTPApi], [EN][v4_EN_HTTPApi]), WebRTC([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]), etc. +* tcp://8080, HTTP live streaming server, HTTP-FLV([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv]), HLS([CN][v4_CN_SampleHLS], [EN][v4_EN_SampleHLS]) as such. +* udp://8000, WebRTC Media([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]) server. + +For optional HTTPS services, which might be provided by other web servers: -* tcp://1935, for RTMP live streaming server. -* tcp://1985, HTTP API server. -* tcp://1990, HTTPS API server. -* tcp://8080, HTTP live streaming server. * tcp://8088, HTTPS live streaming server. -* udp://8000, [WebRTC Media](https://github.com/ossrs/srs/wiki/v4_CN_WebRTC) server. -* udp://1980, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) server. +* tcp://1990, HTTPS API server. + +For optional stream caster services, to push streams to SRS: + * udp://8935, Stream Caster: [Push MPEGTS over UDP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-mpeg-ts-over-udp) server. * tcp://554, Stream Caster: [Push RTSP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-rtsp-to-srs) server. * tcp://8936, Stream Caster: [Push HTTP-FLV](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-http-flv-to-srs) server. @@ -93,6 +100,10 @@ The ports used by SRS: * udp://9000, Stream Caster: [Push GB28181 Media(bundle)](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server. * udp://58200-58300, Stream Caster: [Push GB28181 Media(no-bundle)](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server. * udp://10080, Stream Caster: [Push SRT Media](https://github.com/ossrs/srs/issues/1147#issuecomment-577469119) server. + +For external services to work with SRS: + +* udp://1989, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) server. ## Features @@ -1298,14 +1309,13 @@ Remark: | ......) | | | +----------------------+ | | | MediaSource(2) | | | -| (RTSP,FILE, | | | -| HTTP,HLS, --push-+->- StreamCaster(4) -(rtmp)-+-> SRS | -| Device, | | | +| (MPEGTSoverUDP | | | +| HTTP-FLV, --push-+->- StreamCaster(4) -(rtmp)-+-> SRS | +| GB28181,SRT, | | | | ......) | | | +----------------------+ | | | FFMPEG --push(srt)--+->- SRTModule(5) ---(rtmp)-+-> SRS | +----------------------+----------------------------+----------------+ - ``` Remark: diff --git a/trunk/3rdparty/signaling/www/demos/index.html b/trunk/3rdparty/signaling/www/demos/index.html index c2b617daa..20f2f9a00 100644 --- a/trunk/3rdparty/signaling/www/demos/index.html +++ b/trunk/3rdparty/signaling/www/demos/index.html @@ -19,7 +19,16 @@ let elems = document.getElementsByClassName('srs_demo'); for (var i = 0; i < elems.length; i++) { let elem = elems.item(i); - elem.setAttribute('href', elem.getAttribute('href') + '&room=' + roomName); + + // Use random room. + let href = elem.getAttribute('href') + '&room=' + roomName; + + // For run demos on SRS http server. + if (window.location.port === '8080') { + href += '&wsp=1989'; + } + + elem.setAttribute('href', href); } diff --git a/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js b/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js index 4b030542d..6d4c6a8b4 100644 --- a/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js +++ b/trunk/3rdparty/signaling/www/demos/js/srs.sdk.js @@ -29,6 +29,14 @@ function SrsRtcPublisherAsync() { var self = {}; + // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia + self.constraints = { + audio: true, + video: { + width: {ideal: 320, max: 576} + } + }; + // @see https://github.com/rtcdn/rtcdn-draft // @url The WebRTC url to play with, for example: // webrtc://r.ossrs.net/live/livestream @@ -56,12 +64,14 @@ function SrsRtcPublisherAsync() { self.pc.addTransceiver("audio", {direction: "sendonly"}); self.pc.addTransceiver("video", {direction: "sendonly"}); - var stream = await navigator.mediaDevices.getUserMedia( - {audio: true, video: {width: {max: 320}}} - ); + 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(); @@ -94,9 +104,6 @@ function SrsRtcPublisherAsync() { ); session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; - // Notify about local stream when success. - self.onaddstream && self.onaddstream({stream: stream}); - return session; }; @@ -107,7 +114,10 @@ function SrsRtcPublisherAsync() { }; // The callback when got local stream. - self.onaddstream = function (event) { + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack + self.ontrack = function (event) { + // Add track to stream of SDK. + self.stream.addTrack(event.track); }; // Internal APIs. @@ -253,6 +263,11 @@ function SrsRtcPublisherAsync() { self.pc = new RTCPeerConnection(null); + // To keep api consistent between player and publisher. + // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack + // @see https://webrtc.org/getting-started/media-devices + self.stream = new MediaStream(); + return self; } @@ -315,6 +330,7 @@ function SrsRtcPlayerAsync() { await self.pc.setRemoteDescription( new RTCSessionDescription({type: 'answer', sdp: session.sdp}) ); + session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/'; return session; }; @@ -324,8 +340,12 @@ function SrsRtcPlayerAsync() { self.pc = null; }; - // The callback when got remote stream. - self.onaddstream = function (event) {}; + // The callback when got remote track. + // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream + self.ontrack = function (event) { + // https://webrtc.org/getting-started/remote-streams + self.stream.addTrack(event.track); + }; // Internal APIs. self.__internal = { @@ -469,9 +489,14 @@ function SrsRtcPlayerAsync() { }; self.pc = new RTCPeerConnection(null); - self.pc.onaddstream = function (event) { - if (self.onaddstream) { - self.onaddstream(event); + + // 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); } }; @@ -483,7 +508,8 @@ function SrsRtcPlayerAsync() { function SrsRtcFormatSenders(senders, kind) { var codecs = []; senders.forEach(function (sender) { - sender.getParameters().codecs.forEach(function(c) { + var params = sender.getParameters(); + params && params.codecs && params.codecs.forEach(function(c) { if (kind && sender.track.kind !== kind) { return; } diff --git a/trunk/3rdparty/signaling/www/demos/one2one.html b/trunk/3rdparty/signaling/www/demos/one2one.html index 6271e0511..5bef9a4ba 100644 --- a/trunk/3rdparty/signaling/www/demos/one2one.html +++ b/trunk/3rdparty/signaling/www/demos/one2one.html @@ -22,7 +22,7 @@ SRS