ossrs / srs Goto Github PK

SRS is a simple, high-efficiency, real-time video server supporting RTMP, WebRTC, HLS, HTTP-FLV, SRT, MPEG-DASH, and GB28181.

Home Page: https://ossrs.io

License: MIT License

Shell 2.33% CMake 0.06% Python 0.19% HTML 3.38% C++ 87.47% Makefile 0.03% JavaScript 5.76% Go 0.59% CSS 0.01% Dockerfile 0.03% Lua 0.04% Batchfile 0.01% NSIS 0.09%

rtmp live-streaming live media-server hls webrtc srt low-latency video-conferencing video

srs's Introduction

SRS(Simple Realtime Server)

SRS/6.0 (Hang) is a simple, high-efficiency, and real-time video server, supporting RTMP/WebRTC/HLS/HTTP-FLV/SRT/MPEG-DASH/GB28181, Linux/Windows/macOS, X86_64/ARMv7/AARCH64/M1/RISCV/LOONGARCH/MIPS, and essential features.

Note: For more details on the single-node architecture for SRS, please visit the following link.

SRS is licenced under MIT, and some third-party libraries are distributed under their licenses.

Usage

Please check the Getting Started guide in English or Chinese. We highly recommend using SRS with docker:

docker run --rm -it -p 1935:1935 -p 1985:1985 -p 8080:8080 \
    -p 8000:8000/udp -p 10080:10080/udp ossrs/srs:5

Tips: If you're in China, use this image registry.cn-hangzhou.aliyuncs.com/ossrs/srs:5 for faster speed.

Open http://localhost:8080/ to verify, and then stream using the following FFmpeg command:

ffmpeg -re -i ./doc/source.flv -c copy -f flv -y rtmp://localhost/live/livestream

Alternatively, stream by OBS using the following configuration:

Service: Custom
Server: rtmp://localhost/live
Stream Key: livestream

Play the following streams using media players:

To play an RTMP stream with URL rtmp://localhost/live/livestream on VLC player, open the player, go to Media > Open Network Stream, enter the URL and click Play.
You can play HTTP-FLV stream URL http://localhost:8080/live/livestream.flv on a webpage using the srs-player, an HTML5-based player.
Use srs-player for playing HLS stream with URL http://localhost:8080/live/livestream.m3u8.

If you'd like to use WebRTC, convert RTMP to WebRTC, or convert WebRTC to RTMP, please check out the wiki documentation in either English or Chinese.

To learn more about RTMP, HLS, HTTP-FLV, SRT, MPEG-DASH, WebRTC protocols, clustering, HTTP API, DVR, and transcoding, please check the documents in English or Chinese.

Sponsor

Would you like additional assistance from us? By becoming a sponsor or backer of SRS, we can provide you with the support you need:

Backer: $5 per month, online text chat support through Discord.
Sponsor: $100 per month, online text chat plus online meeting support.

Please visit OpenCollective to become a backer or sponsor, and send us a direct message on Discord. We are currently providing support to the developers listed below:

At SRS, our goal is to create a free, open-source community that helps developers all over the world build high-quality streaming and RTC platforms for their businesses.

Contributing

The authors, TOC(Technical Oversight Committee), and contributors are listed here. The TOC members who made significant contributions and maintained parts of SRS are listed below:

Winlin: Founder of the project, focusing on ST and Issues/PR. Responsible for architecture and maintenance.
ZhaoWenjie: One of the earliest contributors, focusing on HDS and Windows. Has expertise in client technology.
ShiWei: Specializes in SRT and H.265, maintaining SRT and FLV patches for FFmpeg. An expert in codecs and FFmpeg.
XiaoZhihong: Concentrates on WebRTC/QUIC and SRT, with expertise in network QoS. Contributed to ARM on ST and was the original contributor for WebRTC.
WuPengqiang: Focused on H.265, initially contributed to the FFmpeg module in SRS for transcoding AAC with OPUS for WebRTC.
XiaLixin: Specializes in GB28181, with expertise in live streaming and WebRTC.
LiPeng: Concentrates on WebRTC and contributes to memory management and smart pointers.
ChenGuanghua: Focused on WebRTC/QoS and introduced the Asan toolchain to SRS.
ChenHaibo: Specializes in GB28181 and HTTP API, contributing to patches for FFmpeg with WHIP.
ZhangJunqin: Focused on H.265, Prometheus Exporter, and API module.

A huge THANK YOU goes out to:

All the contributors of SRS.
All the friends of SRS who gave big support.
Genes, Mabbott, and Michael Talyanksy for making and sharing State Threads.

We're really thankful to everyone in the community for helping us find bugs and improve the project. To stay in touch and keep helping our community, please check out this guide.

LICENSE

SRS is licenced under MIT, and some third-party libraries are distributed under their licenses.

Releases

2024-02-15, Release v6.0-d4, v6.0-d4, 6.0 dev4, v6.0.113, 167695 lines.
2023-11-19, Release v6.0-d3, v6.0-d3, 6.0 dev3, v6.0.101, 167560 lines.
2023-09-28, Release v6.0-d2, v6.0-d2, 6.0 dev2, v6.0.85, 167509 lines.
2023-08-31, Release v6.0-d1, v6.0-d1, 6.0 dev1, v6.0.72, 167135 lines.
2023-07-09, Release v6.0-d0, v6.0-d0, 6.0 dev0, v6.0.59, 166739 lines.
2024-04-03, Release v5.0-r2, v5.0-r2, 5.0 release2, v5.0.210, 163515 lines.
2024-02-15, Release v5.0-r1, v5.0-r1, 5.0 release1, v5.0.208, 163441 lines.
2023-12-30, Release v5.0-r0, v5.0-r0, 5.0 release0, v5.0.205, 163363 lines.
2023-11-19, Release v5.0-b7, v5.0-b7, 5.0 beta7, v5.0.200, 163305 lines.
2023-10-25, Release v5.0-b6, v5.0-b6, 5.0 beta6, v5.0.195, 163303 lines.
2023-09-28, Release v5.0-b5, v5.0-b5, 5.0 beta5, v5.0.185, 163254 lines.
2023-08-31, Release v5.0-b4, v5.0-b4, 5.0 beta4, v5.0.176, 162919 lines.
2023-08-02, Release v5.0-b3, v5.0-b3, 5.0 beta3, v5.0.170, 162704 lines.
2023-07-09, Release v5.0-b2, v5.0-b2, 5.0 beta2, v5.0.166, 162520 lines.
2023-06-11, Release v5.0-b1, v5.0-b1, 5.0 beta1, v5.0.157, 162494 lines.
2023-05-14, Release v5.0-b0, v5.0-b0, 5.0 beta0, v5.0.155, 162600 lines.
2023-03-23, Release v5.0-a5, v5.0-a5, 5.0 alpha5, v5.0.148, 162066 lines.
2023-02-12, Release v5.0-a4, v5.0-a4, 5.0 alpha4, v5.0.141, 161897 lines.
2023-01-02, Release v5.0-a3, v5.0-a3, 5.0 alpha3, v5.0.128, 161327 lines.
2022-12-18, Release v5.0-a2, v5.0-a2, 5.0 alpha2, v5.0.112, 161233 lines.
2022-12-01, Release v5.0-a1, v5.0-a1, 5.0 alpha1, v5.0.100, 160817 lines.
2022-11-25, Release v5.0-a0, v5.0-a0, 5.0 alpha0, v5.0.98, 159813 lines.
2022-11-22, Release v4.0-r4, v4.0-r4, 4.0 release4, v4.0.268, 145482 lines.
2022-09-16, Release v4.0-r3, v4.0-r3, 4.0 release3, v4.0.265, 145328 lines.
2022-08-24, Release v4.0-r2, v4.0-r2, 4.0 release2, v4.0.257, 144890 lines.
2022-06-29, Release v4.0-r1, v4.0-r1, 4.0 release1, v4.0.253, 144680 lines.
2022-06-11, Release v4.0-r0, v4.0-r0, 4.0 release0, v4.0.252, 144680 lines.
2020-06-27, Release v3.0-r0, 3.0 release0, 3.0.141, 122674 lines.
2020-02-02, Release v3.0-b0, 3.0 beta0, 3.0.112, 121709 lines.
2019-10-04, Release v3.0-a0, 3.0 alpha0, 3.0.56, 107946 lines.
2017-03-03, Release v2.0-r0, 2.0 release0, 2.0.234, 86373 lines.
2016-08-06, Release v2.0-b0, 2.0 beta0, 2.0.210, 89704 lines.
2015-08-23, Release v2.0-a0, 2.0 alpha0, 2.0.185, 89022 lines.
2014-12-05, Release v1.0-r0, all bug fixed, 1.0.10, 59391 lines.
2014-10-09, Release v0.9.8, all bug fixed, 1.0.0, 59316 lines.
2014-04-07, Release v0.9.1, live streaming. 30000 lines.
2013-10-23, Release v0.1.0, rtmp. 8287 lines.
2013-10-17, Created.

Features

Please read FEATURES.

Changelog

Please read CHANGELOG.

Performance

Please read PERFORMANCE.

Architecture

Please read ARCHITECTURE.

Ports

Please read PORTS.

APIs

Please read APIS.

Mirrors

Please read MIRRORS.

Dockers

Please read DOCKERS.

Beijing, 2013.10
Winlin

srs's People

Contributors

Stargazers

Watchers

Forkers

huzhenyu sqbing homelee vmespace seoultsai 5ant yfqian yvlf deluxor msimen firehare sdjc mr8 vinciqin cittu firstblade jrz90000 lingguo jboss42 chengmo03013106 laloli groveme asamqi ytdxcmq90 mruse ninetian yikai2651 wangcrystal jimyong88 backylee a64307410 djtms corefan flashbuckets togear chriszeng87 williamren eebwen synote dymx101 zhangf911 codersadis yefayang ljxie livestudiosolution lvjh bboniao wenqinruan qqzuowei newlightdd juannywang 9612jhf itneste skyfe79 danielfree callmenp otnp airkraken nestle1998 angus1105 arbence zyq2007 sopnic xclouded icevl campones qxh0329 qilongzhang rageyang killheaven luckcape gouravd wljcom qingwa5555 istig lewdon lixiong4788 jiangbing9293 toosgle winlinvip ik3210 yuehonghui ekapujiw2002 houcy smartdu dourgulf wangtingbo noname007 gitter-badger jerryoneal hippocampi cuigl marslabtron ziyan kaiwisdom akfork shenhailuanma icechenbing x5u neoyap

srs's Issues

Reduce the minimum latency of the RTMP server

RTMP Low Latency.

When outputting HLS, support AnnexB format in RTMP packets.

https://github.com/winlinvip/simple-rtmp-server/issues/184

TRANS_BY_GPT3

MP3: Support mp3 for RTMP/HLS/HTTP-FLV/HTTP-TS/HLS etc.

SRS 3/4 supports MP3 over RTMP/HTTP-FLV/HTTP-TS, however there is some bugs need to be fixed.

Usage

Start SRS server:

# For RTMP/HTTP-FLV/HLS
./objs/srs -c conf/mp3.conf

# For HTTP-TS
./objs/srs -c conf/mp3.ts.conf

# For HTTP-MP3
./objs/srs -c conf/mp3.http.conf

# For RTMP(MP3) to WebRTC
./objs/srs -c conf/mp3.rtc.conf

Publish RTMP stream with mp3 audio:

ffmpeg -stream_loop -1  -re -i doc/source.flv -vcodec copy -acodec libmp3lame \
    -f flv rtmp://localhost/live/livestream

Play live stream:

RTMP by VLC or ffplay: rtmp://localhost/live/livestream
HTTP-FLV by VLC, ffplay or H5(srs-player): http://localhost:8080/live/livestream.flv
HTTP-MP3 by VLC, ffplay or H5(srs-player): http://localhost:8080/live/livestream.mp3
HTTP-TS by VLC, ffplay or H5(srs-player): http://localhost:8080/live/livestream.ts
HLS by VLC, ffplay or H5(srs-player): http://localhost:8080/live/livestream.m3u8
Covert RTMP(MP3) to WebRTC, play by H5(srs-player): webrtc://localhost/live/livestream

Status

I tested the ability for MP3 over serval protocols, found that some is ok but others fail, and listed the test result bellow.

Note: There is a bug for playing HTTP-TS with mp3 audio by H5, see mpegts.js #83

One connection does not currently support multiple flow channels

https://github.com/winlinvip/simple-rtmp-server/issues/156

TRANS_BY_GPT3

Support http-mp4 file stream, fmp4 stream.

https://github.com/winlinvip/simple-rtmp-server/issues/174

TRANS_BY_GPT3

DASH: Support convert RTMP/WebRTC to MPEG-DASH

Usage

Make sure your SRS version is SRS 5.0.96+.

Build SRS and run:

./configure && make && ./objs/srs -c conf/dash.conf

Publish by FFmpeg:

ffmpeg -re -i doc/source.flv -c copy -f flv rtmp://localhost/live/livestream

Play by

FFplay ffplay http://localhost:8080/live/livestream.mpd
VLC http://localhost:8080/live/livestream.mpd
Chrome: http://localhost:8080/live/livestream.mpd

The cache of SrsStreamCache causes the consumer to not release.

https://github.com/winlinvip/simple-rtmp-server/issues/352

TRANS_BY_GPT3

Support log splitting, LOG ROTATE

https://github.com/winlinvip/simple-rtmp-server/issues/105

TRANS_BY_GPT3

Can you support a feature similar to "hls_base_url" in nginx-rtmp?

https://github.com/winlinvip/simple-rtmp-server/issues/340

TRANS_BY_GPT3

After the RTMP streaming of the video source stops, can the corresponding HLS files m3u8 and ts be cleaned up? #209

https://github.com/winlinvip/simple-rtmp-server/issues/209

TRANS_BY_GPT3

KERNEL: Software Consistency and Conceptual Integrity Issues

https://github.com/winlinvip/simple-rtmp-server/issues/288

TRANS_BY_GPT3

UDP protocol is used between edge and upnode.

https://github.com/winlinvip/simple-rtmp-server/issues/94

TRANS_BY_GPT3

HLS fault backup, HLS hot standby

https://github.com/winlinvip/simple-rtmp-server/issues/351

TRANS_BY_GPT3

The `st_usleep()` inside the `st` may cause a long hang when the system time is rolled back.

https://github.com/winlinvip/simple-rtmp-server/issues/116

TRANS_BY_GPT3

When the nal_unit_type of HLS is 9, it should be ignored directly.

https://github.com/winlinvip/simple-rtmp-server/issues/281

TRANS_BY_GPT3

HTTP FLV lacks authentication verification mechanism

https://github.com/winlinvip/simple-rtmp-server/issues/345

TRANS_BY_GPT3

Issue with default value of transcoding resolution

https://github.com/winlinvip/simple-rtmp-server/issues/169

TRANS_BY_GPT3

Support Mac OSX

https://github.com/winlinvip/simple-rtmp-server/issues/155

TRANS_BY_GPT3

SRS VOD #205

https://github.com/winlinvip/simple-rtmp-server/issues/205

KERNEL: Correctly using ST threads, an HTTP request shorter than 10 milliseconds may cause a segmentation fault.

https://github.com/winlinvip/simple-rtmp-server/issues/110

TRANS_BY_GPT3

Support RTMP push stream authentication, RTMP authentication.

https://github.com/winlinvip/simple-rtmp-server/issues/71

TRANS_BY_GPT3

The public key generated by openssl is sometimes not 128 bytes.

https://github.com/winlinvip/simple-rtmp-server/issues/165

TRANS_BY_GPT3

haw to Specifies external command with arguments to be executed on play event

https://github.com/winlinvip/simple-rtmp-server/issues/318

If the publisher stops streaming, continuous connections from the client will cause the number of connections (conns) to keep increasing. This will quickly reach the maximum number of connections (max_connections), resulting in new clients being unable to connect.

https://github.com/winlinvip/simple-rtmp-server/issues/161

TRANS_BY_GPT3

HTTP supports vhost

https://github.com/winlinvip/simple-rtmp-server/issues/140

TRANS_BY_GPT3

HTTP URL and Response Architecture Adjustment

https://github.com/winlinvip/simple-rtmp-server/pull/277

TRANS_BY_GPT3

FRSC: Supports sub-millisecond latency, replaceable fast channel technology.

https://github.com/winlinvip/simple-rtmp-server/issues/120

Supports sub-100 millisecond latency, replaceable fast channel technology
Support <500ms latency, FRSC(fast RTMP-compatible stream channel tech)

Sub-100 millisecond latency requires improvement on the existing RTMP mechanism:

Consider using UDP instead of TCP for encoder transmission, server internal communication, and client transmission. Support switchable UDP/TCP mechanism, for example, only the client uses TCP/RTMP while others use UDP private protocols.
Consider non-interlaced media stream transmission: The encoder BLE supports non-interlaced output, where the audio stream timestamp increases monotonically, the video stream increases monotonically, but they are not mixed together. Mixing them would increase latency. The server does not mix them internally (interlaced) either, instead, the mixing of video and audio is done at the edge server or on the client side (own client).
Consider using IO mechanism instead of timeout mechanism: Currently, the server logic waits for 300 milliseconds for client control messages and then sends all pending packets together to the client. This introduces a delay of 300 milliseconds, which increases at each level (according to average probability). If the 300-millisecond timeout mechanism is replaced with an IO mechanism, i.e., the "write" event for packet transmission becomes an IO "read" event, it can wait for two file descriptors to determine whether there is a message from the client or a message to be written.
Packet loss mechanism: Consider supporting packet loss at the edge or in the encoder to control the delay within the sub-500 millisecond range, ideally around 300 milliseconds for Flash clients. If the server controls the delay within 200 milliseconds, the overall system delay can be controlled within 500 milliseconds. If the client is a custom client, the delay can be even lower (server-side control within 200 milliseconds).
Rollback to standard mechanism: Support standard RTMP, for example, within the encoder/server cluster, use low-latency distribution mechanism. If it is detected that the client is Flash-based, the edge should mix the video and audio streams and send them to the client with monotonically increasing timestamps.

In summary, supporting sub-100 millisecond latency requires a comprehensive set of mechanisms and customization on top of RTMP. Compatibility with standard RTMP should also be considered. Therefore, the main goal of the system is to establish a fast channel, control the latency within the sub-100 millisecond range, and support integration with universal solutions at any point in the fast channel.

TRANS_BY_GPT3

MPEGTS write the pcr in right way.

https://github.com/winlinvip/simple-rtmp-server/issues/311

Publish by MPEG-TS over UDP to SRS.

Support MPEGTS over UDP streaming to SRS.

Start SRS.

./objs/srs -c conf/push.mpegts.over.udp.conf

Start FFmpeg streaming.

ffmpeg -re -f flv -i doc/source.flv -c copy -f mpegts udp://127.0.0.1:8935

Play: http://localhost:8080/players/srs_player.html?autostart=true

TRANS_BY_GPT3

DVR file naming rules, supporting stream and date naming, including directory and file names.

https://github.com/winlinvip/simple-rtmp-server/issues/179

TRANS_BY_GPT3

Support AMF3

https://github.com/winlinvip/simple-rtmp-server/issues/131

TRANS_BY_GPT3

HTTP FLV Live Stream triggers edge to pull content from origin

https://github.com/winlinvip/simple-rtmp-server/issues/324

TRANS_BY_GPT3

listen parameter does not support specifying IP address

https://github.com/winlinvip/simple-rtmp-server/issues/149

TRANS_BY_GPT3

Support RTMP source penetration authentication

https://github.com/winlinvip/simple-rtmp-server/issues/104

TRANS_BY_GPT3

Can RTMP stream forwarding be specified through a URL?

Currently, the forwarding mode in circulation needs to be configured through a configuration file. Can you please add the option to configure the forwarding URL?

For example:
Forwarding URL: rtmp://192.168.1.2/myapp/mystream?forward=rtmp://cdn.com/myapp/mystream?XXXXXX

This would make it more convenient to change whether the stream needs to be forwarded at any time. Currently, we have a use case where most of the streams only need to be viewed within the local network and do not require internet access. However, sometimes certain playback sources need to be played on the internet. Configuring this forwarding through a config file is not very flexible.

Could you please add this functionality or suggest a way to achieve a similar effect? Alternatively, could you provide guidance on how I can quickly modify the code to achieve this functionality?

TRANS_BY_GPT3

HLS: Join no desc to solve the problem with Samsung phones.

https://github.com/winlinvip/simple-rtmp-server/pull/114

TRANS_BY_GPT3

WebRTC: Support WebRTC, for video meeting, SFU server, video converence.

WebRTC is now quite mature, with stable playback and the protocol already being an RFC. There are also quite a few corresponding open-source projects. However, I believe that WebRTC still lacks a high-performance, simple and easy-to-use server. I have analyzed the existing servers before and found various issues. SRS has a great opportunity to solve these problems.

See English or Chinese

Other than the ability to reload, all other RAW API capabilities have been removed.

Provide an HTTP API to modify the configuration file and then reload the effective configuration.
https://github.com/winlinvip/simple-rtmp-server/issues/319

TRANS_BY_GPT3

Delivery Adobe HDS

https://github.com/winlinvip/simple-rtmp-server/issues/328

Multiple processes use REUSEADDR without using master

https://github.com/winlinvip/simple-rtmp-server/issues/170

TRANS_BY_GPT3

ossrs / srs Goto Github PK

srs's Introduction

SRS(Simple Realtime Server)

Usage

Sponsor

Contributing

LICENSE

Releases

Features

Changelog

Performance

Architecture

Ports

APIs

Mirrors

Dockers

srs's People

Contributors

Stargazers

Watchers

Forkers

srs's Issues

Usage

Status

Usage

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