SRS(Simple Realtime Server)

SRS/8.0 (Free) is a simple, high-performance, AI-driven real-time media server. It supports RTMP, WebRTC, HLS, HTTP-FLV, SRT, MPEG-DASH, and GB28181 on Linux and macOS across x86_64, ARMv7, AArch64, Apple M1, RISC-V, LoongArch, and MIPS architectures, and natively on Windows x64. Supported codecs include H.264, H.265, AV1, VP9, AAC, Opus, and G.711. See the complete list of 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:6

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.

AI Agent

I recommend to use AI to understand and maintain your SRS, please follow the wiki AI Agent for details.

Contributing

The maintainers, and contributors are listed here. The maintainers who made significant contributions and maintained parts of SRS are listed below, ranked by the number of commits:

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

This is the beginning of the README. Read the rest at the original. Read the full README ↗