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SRS is a simple, high-efficiency, real-time media server supporting RTMP, WebRTC, HLS, HTTP-FLV, HTTP-TS, SRT, MPEG-DASH, and GB28181, with codec support for H.264, H.265, AV1, VP9, AAC, Opus, and G.711.
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RTC: audio packet jitter buffer. v7.0.48 (#4295) 
Rtp packets may be retransmitted, disordered, jittery, delayed,
etc.There may be abnormalities when converting to rtmp.

To reproduce this problem, you need to set the network reordering by
[tc-ui](https://github.com/ossrs/tc-ui). Note that you need a linux
server, and start it by docker:

```bash
docker run --network=host --privileged -it --restart always -d \
    --name tc -v /lib/modules:/lib/modules:ro ossrs/tc-ui:1
```

Set up 5% packet reordering and a 1ms delay; then you will notice that
the audio is stuttering, somewhat noisy, and lacks fluency.

```bash
curl http://localhost:2023/tc/api/v1/config/raw -X POST \
  -d 'tcset ens5 --direction incoming --delay 40ms --reordering 5% --port 8000'
```

> Note: Even without network conditions, the natural state can also
cause packet reordering, especially in public cloud platforms such as
AWS EC2.

> Note: You can use command `curl
http://localhost:2023/tc/api/v1/config/raw -X POST -d 'tcdel --all
ens5'` to reset the network condition settings.

Check the web console, you will see the reordering setup:

<img width="500" alt="TC Settings"
src="https://github.com/user-attachments/assets/b278fdf4-9fcc-4aac-b534-dfa34e28c371"
/>

Then, publish stream via WHIP: http://localhost:8080/players/whip.html

And, play via HTTP-FLV: http://localhost:8080/players/srs_player.html

Finished by AI:

* [AI: Extract audio jitter buffer to class
AudioPacketCache](a4097d9374)
* [AI: Add utest and fix
bug.](c919227af5)

---------

Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: winlin <winlinvip@gmail.com>
2025-07-16 21:36:56 -04:00
.github Update bug_report.md 2025-07-10 08:00:26 -04:00
.run Heartbeat: Report ports for proxy server. v5.0.215 v6.0.156 v7.0.15 (#4171) 2024-09-09 10:37:41 +08:00
.vscode AI: Add run utest in guideline for augment code. 2025-06-30 08:10:21 -04:00
proxy Migrate proxy to ossrs/proxy-go repository. 2025-06-23 10:39:57 -04:00
trunk RTC: audio packet jitter buffer. v7.0.48 (#4295) 2025-07-16 21:36:56 -04:00
.augment-guidelines AI: Update time guideline for augment. 2025-07-12 21:36:47 -04:00
.augmentignore AI: Update enhanced-rtmp spec for augment code. 2025-07-04 08:39:23 -04:00
.clang-format Script: Use clang-format to unify the coding style. v7.0.38 (#4366) 2025-06-01 22:01:15 -04:00
.gitignore rtc2rtmp: Support RTC-to-RTMP remuxing with HEVC. v7.0.43 (#4349) 2025-07-03 08:24:42 -04:00
CONTRIBUTING.md Update CHANGELOG for #4309. v7.0.31 2025-04-26 06:58:00 -04:00
Dockerfile Dockerfile: Consistently use proper ENV syntax using equal sign. v6.0.140 (#4116) 2024-07-24 10:19:05 +08:00
LICENSE update copyright to 2025. v5.0.218 v6.0.159 v7.0.21 (#4271) 2025-01-14 17:35:18 +08:00
README.md Update the release in the README for consistent. v7.0.40 (#4341) 2025-06-04 08:08:00 -04:00
SECURITY.md Update SECURITY policy. 2023-01-30 14:20:37 +08:00

README.md

SRS(Simple Realtime Server)

SRS/7.0 (Kai) 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.

SRS Overview

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:

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.

If you want to use an IDE, VSCode is recommanded. VSCode supports macOS, Linux, and Windows platforms. The settings are ready. All you need to do is open the folder with VSCode and enjoy the efficiency brought by the IDE. See VSCode README for details.

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:

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

FOSSA Status

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

Releases

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