From eba78149adc350ecc4e13fe4d56530cb4b3eeca6 Mon Sep 17 00:00:00 2001 From: winlin Date: Sat, 24 Apr 2021 21:40:09 +0800 Subject: [PATCH 1/2] SquashSRS4: Change push-RTSP as deprecated feature. --- README.md | 3 ++- trunk/conf/full.conf | 1 + trunk/src/app/srs_app_server.cpp | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/README.md b/README.md index f7836a048..263808b80 100755 --- a/README.md +++ b/README.md @@ -166,10 +166,10 @@ Other documents: - [x] [Experimental] Support DVR in MP4 format, read [#738][bug #738]. - [x] [Experimental] Support MPEG-DASH, the future live streaming protocol, read [#299][bug #299]. - [x] [Experimental] Support pushing MPEG-TS over UDP, please read [bug #250][bug #250]. -- [x] [Experimental] Support pushing RTSP, please read [bug #133][bug #133]. - [x] [Experimental] Support pushing FLV over HTTP POST, please read wiki([CN][v2_CN_Streamer2], [EN][v2_EN_Streamer2]). - [x] [Experimental] Support HTTP RAW API, please read [#459][bug #459], [#470][bug #470], [#319][bug #319]. - [x] [Experimental] Support SRT server, read [#1147][bug #1147]. +- [x] [Deprecated] Support pushing RTSP, please read [bug #133][bug #133]. - [x] [Deprecated] Support RTMP client library: srs-librtmp([CN][v3_CN_SrsLibrtmp], [EN][v3_EN_SrsLibrtmp]) - [x] [Deprecated] Support Adobe HDS(f4m), please read wiki([CN][v2_CN_DeliveryHDS], [EN][v2_EN_DeliveryHDS]) and [#1535][bug #1535]. - [x] [Deprecated] Support bandwidth testing([CN][v1_CN_BandwidthTestTool], [EN][v1_EN_BandwidthTestTool]), please read [#1535][bug #1535]. @@ -197,6 +197,7 @@ Other documents: ## V4 changes +* v4.0, 2021-04-24, Change push-RTSP as deprecated feature. * v4.0, 2021-04-24, Player: Change the default from RTMP to HTTP-FLV. * v4.0, 2021-04-24, Disable CherryPy by --cherrypy=off. 4.0.90 * v4.0, 2021-04-01, RTC: Refine TWCC and SDP exchange. 4.0.88 diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf index 70d199e4a..b17085f03 100644 --- a/trunk/conf/full.conf +++ b/trunk/conf/full.conf @@ -288,6 +288,7 @@ stream_caster { } # RTSP +# It's deprecated and will be removed in the future, see [#2304](https://github.com/ossrs/srs/issues/2304#issuecomment-826009290). stream_caster { # whether stream caster is enabled. # default: off diff --git a/trunk/src/app/srs_app_server.cpp b/trunk/src/app/srs_app_server.cpp index ceda17b03..da93d1c4f 100644 --- a/trunk/src/app/srs_app_server.cpp +++ b/trunk/src/app/srs_app_server.cpp @@ -1482,6 +1482,7 @@ srs_error_t SrsServer::listen_stream_caster() if (srs_stream_caster_is_udp(caster)) { listener = new SrsUdpCasterListener(this, SrsListenerMpegTsOverUdp, stream_caster); } else if (srs_stream_caster_is_rtsp(caster)) { + srs_warn("It's deprecated and will be removed in the future, see https://github.com/ossrs/srs/issues/2304#issuecomment-826009290"); listener = new SrsRtspListener(this, SrsListenerRtsp, stream_caster); } else if (srs_stream_caster_is_flv(caster)) { listener = new SrsHttpFlvListener(this, SrsListenerFlv, stream_caster); From fe1c3a4ce13a27e2f2dbd497592f64f9326c714d Mon Sep 17 00:00:00 2001 From: winlin Date: Sat, 24 Apr 2021 22:52:48 +0800 Subject: [PATCH 2/2] SquashSRS4: Update readme --- README.md | 62 ++++++++++++++++++------------------------------------- 1 file changed, 20 insertions(+), 42 deletions(-) diff --git a/README.md b/README.md index 263808b80..f67cfa179 100755 --- a/README.md +++ b/README.md @@ -14,75 +14,42 @@ SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC ## Usage -Recommend to run SRS by [docker][docker-srs4]: +Recommend running 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): ```bash -docker run --rm -p 1935:1935 -p 1985:1985 -p 8080:8080 \ - ossrs/srs:v4.0.76 - -# Or, for developers in China to speedup. -docker run --rm -p 1935:1935 -p 1985:1985 -p 8080:8080 \ - registry.cn-hangzhou.aliyuncs.com/ossrs/srs:v4.0.76 - -# For macOS to enable WebRTC, other OS please see #307. docker run --rm -p 1935:1935 -p 1985:1985 -p 8080:8080 \ --env CANDIDATE=$(ifconfig en0 inet| grep 'inet '|awk '{print $2}') -p 8000:8000/udp \ - registry.cn-hangzhou.aliyuncs.com/ossrs/srs:v4.0.76 + ossrs/srs:v4.0.85 ``` -> Note: All [tags](https://github.com/ossrs/srs/tags) are available, such as -> `ossrs/srs:v3.0-r3` for tag [v3.0-r3](https://github.com/ossrs/srs/releases/tag/v3.0-r3), -> please check at [here](https://cr.console.aliyun.com/repository/cn-hangzhou/ossrs/srs/images) -> or [there](https://hub.docker.com/r/ossrs/srs/tags). - > To enable WebRTC, user MUST set the env `CANDIDATE`, see [#307](https://github.com/ossrs/srs/issues/307#issue-76908382). -If it works, open [http://localhost:8080/](http://localhost:8080/) to check it, then publish +Open [http://localhost:8080/](http://localhost:8080/) to check it, then publish [stream](https://github.com/ossrs/srs/blob/3.0release/trunk/doc/source.200kbps.768x320.flv) by: ```bash -ffmpeg -re -i doc/source.200kbps.768x320.flv -c copy \ - -f flv rtmp://localhost/live/livestream - -# Or by FFmpeg docker -docker run --rm --network=host registry.cn-hangzhou.aliyuncs.com/ossrs/srs:encoder \ - ffmpeg -re -i ./doc/source.200kbps.768x320.flv -c copy \ - -f flv -y rtmp://localhost/live/livestream +docker run --rm --network=host ossrs/srs:encoder ffmpeg -re -i ./doc/source.200kbps.768x320.flv \ + -c copy -f flv -y rtmp://localhost/live/livestream ``` - > Note: If WebRTC enabled, you can publish by [H5](http://localhost:8080/players/rtc_publisher.html?autostart=true). -Play the following streams by players: +Play the following streams by [players](https://ossrs.net): * VLC(RTMP): rtmp://localhost/live/livestream * H5(HTTP-FLV): [http://localhost:8080/live/livestream.flv](http://localhost:8080/players/srs_player.html?autostart=true&stream=livestream.flv&port=8080&schema=http) * H5(HLS): [http://localhost:8080/live/livestream.m3u8](http://localhost:8080/players/srs_player.html?autostart=true&stream=livestream.m3u8&port=8080&schema=http) * H5(WebRTC): [webrtc://localhost/live/livestream](http://localhost:8080/players/rtc_player.html?autostart=true) -> The online demos and players are available on [ossrs.net](https://ossrs.net). - -Strongly recommend reading bellow wikis: - -* How to deliver RTMP streaming?([CN][v1_CN_SampleRTMP], [EN][v1_EN_SampleRTMP]) -* How to build RTMP Edge-Cluster?([CN][v3_CN_SampleRTMPCluster], [EN][v3_EN_SampleRTMPCluster]) -* How to build RTMP Origin-Cluster?([CN][v3_CN_SampleOriginCluster], [EN][v3_EN_SampleOriginCluster]) -* How to deliver HTTP-FLV streaming?([CN][v3_CN_SampleHttpFlv], [EN][v3_EN_SampleHttpFlv]) -* How to deliver HLS streaming?([CN][v3_CN_SampleHLS], [EN][v3_EN_SampleHLS]) -* How to deliver low-latency streaming?([CN][v3_CN_SampleRealtime], [EN][v3_EN_SampleRealtime]) -* Usage: How to play WebRTC from SRS? [#307](https://github.com/ossrs/srs/issues/307) -* Usage: How to publish WebRTC to SRS? [#307](https://github.com/ossrs/srs/issues/307) - It's also very easy to build from source: **>>> Step 1:** Get SRS. ``` -git clone https://gitee.com/winlinvip/srs.oschina.git srs && -cd srs/trunk && git remote set-url origin https://github.com/ossrs/srs.git && -git checkout develop && git pull +git clone -b develop https://gitee.com/winlinvip/srs.oschina.git srs && +cd srs/trunk && git remote set-url origin https://github.com/ossrs/srs.git ``` -> Note: We use [mirrors(gitee)](#mirrors) here, but it's also ok to directly clone by `git clone https://github.com/ossrs/srs.git && cd srs/trunk` +> Note: We use [mirrors(gitee)](#mirrors) here, but it's also ok to `git clone https://github.com/ossrs/srs.git` **>>> Step 2:** Build SRS. @@ -100,6 +67,17 @@ git checkout develop && git pull ./objs/srs -c conf/srs.conf ``` +Strongly recommend reading bellow wikis: + +* How to deliver RTMP streaming?([CN][v1_CN_SampleRTMP], [EN][v1_EN_SampleRTMP]) +* How to build RTMP Edge-Cluster?([CN][v3_CN_SampleRTMPCluster], [EN][v3_EN_SampleRTMPCluster]) +* How to build RTMP Origin-Cluster?([CN][v3_CN_SampleOriginCluster], [EN][v3_EN_SampleOriginCluster]) +* How to deliver HTTP-FLV streaming?([CN][v3_CN_SampleHttpFlv], [EN][v3_EN_SampleHttpFlv]) +* How to deliver HLS streaming?([CN][v3_CN_SampleHLS], [EN][v3_EN_SampleHLS]) +* How to deliver low-latency streaming?([CN][v3_CN_SampleRealtime], [EN][v3_EN_SampleRealtime]) +* Usage: How to play WebRTC from SRS? [#307](https://github.com/ossrs/srs/issues/307) +* Usage: How to publish WebRTC to SRS? [#307](https://github.com/ossrs/srs/issues/307) + Other documents: