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Squash: Fix rtc to rtmp sync timestamp using sender report. #2470
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winlinvip committed Aug 16, 2021
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219 changes: 18 additions & 201 deletions README.md
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[![](https://github.com/ossrs/srs/actions/workflows/release.yml/badge.svg)](https://github.com/ossrs/srs/actions/workflows/release.yml?query=workflow%3ARelease)
[![](https://github.com/ossrs/srs/actions/workflows/test.yml/badge.svg?branch=develop)](https://github.com/ossrs/srs/actions?query=workflow%3ATest+branch%3Adevelop)
[![](https://codecov.io/gh/ossrs/srs/branch/develop/graph/badge.svg)](https://codecov.io/gh/ossrs/srs/branch/develop)
[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/wechat-badge.png)](../../wikis/Contact#wechat)
[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/wechat-badge2.png)](../../wikis/Contact#wechat)
[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/bbs2.png)](http://bbs.ossrs.net)

SRS/4.0 [Leo](https://github.com/ossrs/srs/wiki/v4_CN_Product#release40) 是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT。
Expand Down Expand Up @@ -89,87 +89,9 @@ A big `THANK YOU` also goes to:
* All friends of SRS for [big supports](https://github.com/ossrs/srs/wiki/Product).
* [Genes](http://sourceforge.net/users/genes), [Mabbott](http://sourceforge.net/users/mabbott) and [Michael Talyanksy](https://github.com/michaeltalyansky) for [st](https://github.com/ossrs/state-threads/tree/srs).

## Features

- [x] Using coroutine by ST, it's really simple and stupid enough.
- [x] Support cluster which consists of origin ([CN][v4_CN_DeliveryRTMP],[EN][v4_EN_DeliveryRTMP]) and edge([CN][v4_CN_Edge], [EN][v4_EN_Edge]) server and uses RTMP as default transport protocol.
- [x] Origin server supports remuxing RTMP to HTTP-FLV([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv]) and HLS([CN][v4_CN_DeliveryHLS], [EN][v4_EN_DeliveryHLS]).
- [x] Edge server supports remuxing RTMP to HTTP-FLV([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv]). As for HLS([CN][v4_CN_DeliveryHLS], [EN][v4_EN_DeliveryHLS]) edge server, recomment to use HTTP edge server, such as [NGINX](http://nginx.org/).
- [x] Support HLS with audio-only([CN][v4_CN_DeliveryHLS2], [EN][v4_EN_DeliveryHLS2]), which need to build the timestamp from AAC samples, so we enhanced it please read [#547][bug #547].
- [x] Support HLS with mp3(h.264+mp3) audio codec, please read [bug #301][bug #301].
- [x] Support transmux RTMP to HTTP-FLV/MP3/AAC/TS, please read wiki([CN][v4_CN_DeliveryHttpStream], [EN][v4_CN_DeliveryHttpStream]).
- [x] Support ingesting([CN][v4_CN_Ingest], [EN][v4_EN_Ingest]) other protocols to SRS by FFMPEG.
- [x] Support RTMP long time(>4.6hours) publishing/playing, with the timestamp corrected.
- [x] Support native HTTP server([CN][v4_CN_SampleHTTP], [EN][v4_EN_SampleHTTP]) for http api and http live streaming.
- [x] Support HTTP CORS for js in http api and http live streaming.
- [x] Support HTTP API([CN][v4_CN_HTTPApi], [EN][v4_EN_HTTPApi]) for system management.
- [x] Support HTTP callback([CN][v4_CN_HTTPCallback], [EN][v4_EN_HTTPCallback]) for authentication and integration.
- [x] Support DVR([CN][v4_CN_DVR], [EN][v4_EN_DVR]) to record live streaming to FLV file.
- [x] Support DVR control module like NGINX-RTMP, please read [#459][bug #459].
- [x] Support EXEC like NGINX-RTMP, please read [bug #367][bug #367].
- [x] Support security strategy including allow/deny publish/play IP([CN][v4_CN_Security], [EN][v4_EN_Security]).
- [x] Support low latency(0.1s+) transport model, please read [bug #257][bug #257].
- [x] Support gop-cache([CN][v4_CN_LowLatency2], [EN][v4_EN_LowLatency2]) for player fast startup.
- [x] Support Vhost([CN][v4_CN_RtmpUrlVhost], [EN][v4_EN_RtmpUrlVhost]) and \_\_defaultVhost\_\_.
- [x] Support reloading([CN][v4_CN_Reload], [EN][v4_EN_Reload]) to apply changes of config.
- [x] Support listening at multiple ports.
- [x] Support forwarding([CN][v4_CN_Forward], [EN][v4_EN_Forward]) to other RTMP servers.
- [x] Support transcoding([CN][v4_CN_FFMPEG], [EN][v4_EN_FFMPEG]) by FFMPEG.
- [x] All wikis are writen in [Chinese][v4_CN_Home] and [English][v4_EN_Home].
- [x] Enhanced json, replace NXJSON(LGPL) with json-parser(BSD), read [#904][bug #904].
- [x] Support valgrind and latest ARM by patching ST, read [ST#1](https://github.com/ossrs/state-threads/issues/1) and [ST#2](https://github.com/ossrs/state-threads/issues/2).
- [x] Support traceable and session-based log([CN][v4_CN_SrsLog], [EN][v4_EN_SrsLog]).
- [x] High performance([CN][v4_CN_Performance], [EN][v4_EN_Performance]) RTMP/HTTP-FLV, 6000+ connections.
- [x] Enhanced complex error code with description and stack, read [#913][bug #913].
- [x] Enhanced RTMP url which supports vhost in stream, read [#1059][bug #1059].
- [x] Support origin cluster, please read [#464][bug #464], [RTMP 302][bug #92].
- [x] Support listen at IPv4 and IPv6, read [#460][bug #460].
- [x] Improve test coverage for core/kernel/protocol/service.
- [x] Support docker by [srs-docker](https://github.com/ossrs/srs-docker).
- [x] Support multiple processes by ReusePort([CN][v4_CN_REUSEPORT], [EN][v4_EN_REUSEPORT]), [#775][bug #775].
- [x] Support a simple [mgmt console](http://ossrs.net:8080/console), please read [srs-console](https://github.com/ossrs/srs-console).
- [x] [Experimental] Support playing stream by WebRTC, [#307][bug #307].
- [x] [Experimental] Support publishing stream by WebRTC, [#307][bug #307].
- [x] [Experimental] Support mux RTP/RTCP/DTLS/SRTP on one port for WebRTC, [#307][bug #307].
- [x] [Experimental] Support client address changing for WebRTC, [#307][bug #307].
- [x] [Experimental] Support transcode RTMP/AAC to WebRTC/Opus, [#307][bug #307].
- [x] [Experimental] Support AV1 codec for WebRTC, [#2324][bug #2324].
- [x] [Experimental] Enhance HTTP Stream Server for HTTP-FLV, HTTPS, HLS etc. [#1657][bug #1657].
- [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 FLV over HTTP POST, please read wiki([CN][v4_CN_Streamer2], [EN][v4_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] [Experimental] Support transmux RTC to RTMP, [#2093][bug #2093].
- [x] [Deprecated] Support pushing RTSP, please read [bug #2304][bug #2304].
- [x] [Deprecated] Support Adobe HDS(f4m), please read wiki([CN][v4_CN_DeliveryHDS], [EN][v4_EN_DeliveryHDS]) and [#1535][bug #1535].
- [x] [Deprecated] Support bandwidth testing, please read [#1535][bug #1535].
- [x] [Deprecated] Support Adobe FMS/AMS token traverse([CN][v4_CN_DRM2], [EN][v4_EN_DRM2]) authentication, please read [#1535][bug #1535].
- [x] [Removed] Support RTMP client library: [srs-librtmp](https://github.com/ossrs/srs-librtmp).
- [ ] Support Windows/Cygwin 64bits, [#2532](https://github.com/ossrs/srs/issues/2532).
- [ ] Support push stream by GB28181, [#1500][bug #1500].
- [ ] Support IETF-QUIC for WebRTC Cluster, [#2091][bug #2091].
- [ ] Enhanced forwarding with vhost and variables, [#1342][bug #1342].
- [ ] Support DVR to Cloud Storage, [#1193][bug #1193].
- [ ] Support H.265 over RTMP and HLS, [#465][bug #465].
- [ ] Improve RTC performance to 5K by multiple threading, [#2188][bug #2188].
- [ ] Support source cleanup for idle streams, [#413][bug #413].
- [ ] Support change user to run SRS, [#1111][bug #1111].
- [ ] Support HLS variant, [#463][bug #463].

> Remark: About the milestone and product plan, please read ([CN][v4_CN_Product], [EN][v4_EN_Product]) wiki.
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## Changelog

Please read [CHANGELOG](CHANGELOG.md#changelog).

## Releases

* 2020-08-15, Release [v4.0.156](https://github.com/ossrs/srs/releases/tag/v4.0.156), 4.0 dev4, v4.0.156, 145490 lines.
* 2020-08-14, Release [v4.0.153](https://github.com/ossrs/srs/releases/tag/v4.0.153), 4.0 dev3, v4.0.153, 145506 lines.
* 2020-08-07, Release [v4.0.150](https://github.com/ossrs/srs/releases/tag/v4.0.150), 4.0 dev2, v4.0.150, 145289 lines.
* 2020-07-25, Release [v4.0.146](https://github.com/ossrs/srs/releases/tag/v4.0.146), 4.0 dev1, v4.0.146, 144026 lines.
Expand All @@ -186,146 +108,41 @@ Please read [CHANGELOG](CHANGELOG.md#changelog).
* 2013-10-23, [Release v0.1.0][r0.1], rtmp. 8287 lines.
* 2013-10-17, Created.

## Features

Please read [FEATURES](trunk/doc/Features.md#features).

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## Changelog

Please read [CHANGELOG](trunk/doc/CHANGELOG.md#changelog).

## Compare

Comparing with other media servers, SRS is much better and stronger, for details please
read Product([CN][v4_CN_Compare]/[EN][v4_EN_Compare]).

## Performance

Please read [PERFORMANCE](PERFORMANCE.md#performance).
Please read [PERFORMANCE](trunk/doc/PERFORMANCE.md#performance).

## Architecture

The stream architecture of SRS.

```
+----------+ +----------+
| Upstream | | Deliver |
+---|------+ +----|-----+
+---+------------------+------+---------------------+----------------+
| Input | SRS(Simple RTMP Server) | Output |
+----------------------+----------------------------+----------------+
| | +-> DASH ----------------+-> DASH player |
| Encoder(1) | +-> RTMP/HDS -----------+-> Flash player |
| (FMLE,OBS, --RTMP-+->-+-> HLS/HTTP ------------+-> M3U8 player |
| FFmpeg,XSPLIT, | +-> FLV/MP3/Aac/Ts ------+-> HTTP player |
| ......) | +-> Fowarder ------------+-> RTMP server |
| | +-> Transcoder ----------+-> RTMP server |
| | +-> EXEC(5) -------------+-> External app |
| | +-> DVR -----------------+-> FLV file |
| | +-> BandwidthTest -------+-> Flash |
| | +-> WebRTC --------------+-> Flash |
+----------------------+ | |
| WebRTC Client | +--> RTMP | |
| (H5,Native...) --RTC-+---+---> WebRTC ------------+-> WebRTC Client|
+----------------------+ | |
| MediaSource(2) | | |
| (RTSP,FILE, | | |
| HTTP,HLS, --pull-+->-- Ingester(3) -(rtmp)----+-> SRS |
| Device, | | |
| ......) | | |
+----------------------+ | |
| MediaSource(2) | | |
| (MPEGTSoverUDP | | |
| HTTP-FLV, --push-+->- StreamCaster(4) -(rtmp)-+-> SRS |
| SRT, | | |
| ......) | | |
+----------------------+ | |
| FFMPEG --push(srt)--+->- SRTModule(5) ---(rtmp)-+-> SRS |
+----------------------+----------------------------+----------------+
```

Remark:

1. Encoder: Encoder pushs RTMP stream to SRS.
1. MediaSource: Supports any media source, ingesting by ffmpeg.
1. Ingester: Forks a ffmpeg(or other tools) to ingest as rtmp to SRS, please read [Ingest][v4_CN_Ingest].
1. Streamer: Remuxs other protocols to RTMP, please read [Streamer][v4_CN_Streamer].
1. EXEC: Like NGINX-RTMP, EXEC forks external tools for events, please read [ng-exec][v4_CN_NgExec].
1. SRTModule: A isolate module which run in [hybrid](https://github.com/ossrs/srs/issues/1147#issuecomment-577574883) model.
Please read [ARCHITECTURE](trunk/doc/Architecture.md#architecture).

## Ports

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://8088`, HTTPS live streaming 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.
* `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.
Please read [PORTS](trunk/doc/Resources.md#ports).

## APIs

The API used by SRS:

* `/api/v1/` The HTTP API path.
* `/rtc/v1/` The HTTP API path for RTC.
* `/sig/v1/` The [demo signaling](https://github.com/ossrs/signaling) API.

Other API used by [ossrs.net](https://ossrs.net):

* `/gif/v1` The statistic API.
* `/service/v1/` The latest available version API.
* `/ws-service/v1/` The latest available version API, by websocket.
* `/im-service/v1/` The latest available version API, by IM.
* `/code-service/v1/` The latest available version API, by Code verification.
Please read [APIS](trunk/doc/Resources.md#apis).

## Mirrors

Gitee: [https://gitee.com/ossrs/srs](https://gitee.com/ossrs/srs), the GIT usage([CN][v4_CN_Git], [EN][v4_EN_Git])

```
git clone https://gitee.com/ossrs/srs.git &&
cd srs && git remote set-url origin https://github.com/ossrs/srs.git && git pull
```

> Remark: For users in China, recomment to use mirror from CSDN or OSChina, because they are much faster.
Gitlab: [https://gitlab.com/winlinvip/srs-gitlab](https://gitlab.com/winlinvip/srs-gitlab), the GIT usage([CN][v4_CN_Git], [EN][v4_EN_Git])

```
git clone https://gitlab.com/winlinvip/srs-gitlab.git srs &&
cd srs && git remote set-url origin https://github.com/ossrs/srs.git && git pull
```

Github: [https://github.com/ossrs/srs](https://github.com/ossrs/srs), the GIT usage([CN][v4_CN_Git], [EN][v4_EN_Git])

```
git clone https://github.com/ossrs/srs.git
```

| Branch | Cost | Size | CMD |
| --- | --- | --- | --- |
| 3.0release | 2m19.931s | 262MB | git clone -b 3.0release https://gitee.com/ossrs/srs.git |
| 3.0release | 0m56.515s | 95MB | git clone -b 3.0release --depth=1 https://gitee.com/ossrs/srs.git |
| develop | 2m22.430s | 234MB | git clone -b develop https://gitee.com/ossrs/srs.git |
| develop | 0m46.421s | 42MB | git clone -b develop --depth=1 https://gitee.com/ossrs/srs.git |
| min | 2m22.865s | 217MB | git clone -b min https://gitee.com/ossrs/srs.git |
| min | 0m36.472s | 11MB | git clone -b min --depth=1 https://gitee.com/ossrs/srs.git |

## System Requirements

Supported operating systems and hardware:

* Linux, with x86, x86-64 or arm.
* Mac, with intel chip.
* Other OS, such as Windows, please use [docker](https://github.com/ossrs/srs-docker/tree/v4#usage).
Please read [MIRRORS](trunk/doc/Resources.md#mirrors).

Beijing, 2013.10<br/>
Winlin
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6 changes: 4 additions & 2 deletions trunk/3rdparty/srs-bench/go.mod
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require (
github.com/ossrs/go-oryx-lib v0.0.8
github.com/pion/interceptor v0.0.9
github.com/pion/interceptor v0.0.10
github.com/pion/logging v0.2.2
github.com/pion/rtcp v1.2.6
github.com/pion/rtp v1.6.2
github.com/pion/sdp/v3 v3.0.4
github.com/pion/webrtc/v3 v3.0.4
github.com/pion/transport v0.12.2
github.com/pion/webrtc/v3 v3.0.13
)

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