> Note: For more details on the single-node architecture for SRS, please visit the following [link](https://www.figma.com/file/333POxVznQ8Wz1Rxlppn36/SRS-4.0-Server-Arch).
Please refer to the [Getting Started](https://ossrs.io/lts/en-us/docs/v4/doc/getting-started) or [中文文档：起步](https://ossrs.net/lts/zh-cn/docs/v4/doc/getting-started) guide.
* To play an RTMP stream with URL `rtmp://localhost/live/livestream` on [VLC player](https://www.videolan.org/), 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](http://localhost:8080/players/srs_player.html?autostart=true&stream=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](http://localhost:8080/players/srs_player.html?autostart=true&stream=livestream.m3u8).
* Use the srs-player to play the WebRTC stream with URL [http://localhost:1985/rtc/v1/whip-play/?app=live&stream=livestream](http://localhost:8080/players/whep.html?autostart=true) via WHEP.
> Note: In addition to FFmpeg or OBS, it is possible to [publish by H5](http://localhost:8080/players/whip.html) via WHIP as well.
> To enable WebRTC to publish and convert it to RTMP, please refer to the wiki([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/webrtc#rtc-to-rtmp)) documentation.
> It is essential to ensure the candidate([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate) or [EN](https://ossrs.io/lts/en-us/docs/v4/doc/webrtc#config-candidate))
> is set correctly for WebRTC to avoid potential issues, as it can cause significant problems.
> Note: It is highly recommended to run SRS directly with docker([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/getting-started) / [EN](https://ossrs.io/lts/en-us/docs/v4/doc/getting-started)),
> Note: If you require HTTPS for WebRTC and modern browsers, please refer to the HTTPS API([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/http-api#https-api) / [EN](https://ossrs.io/lts/en-us/docs/v4/doc/http-api#https-api)),
> and HTTPS Live Streaming([CN](https://ossrs.io/lts/en-us/docs/v4/doc/delivery-http-flv#https-flv-live-stream) / [EN](https://ossrs.io/lts/en-us/docs/v4/doc/delivery-http-flv#https-flv-live-stream))
> documentation. Additionally, SRS works perfectly with an HTTPS proxy like Nginx.
Please refer to the following wikis for more information:
* What are the steps to deliver RTMP streaming? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-rtmp), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-rtmp))
* What is the process for delivering WebRTC streaming? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/webrtc))
* What are the steps to convert RTMP to HTTP-FLV streaming? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-http-flv), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-http-flv))
* How can RTMP be converted to HLS streaming? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-hls), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-hls))
* What is the best approach for delivering low-latency streaming? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-realtime), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-realtime))
* How can an RTMP Edge-Cluster be constructed? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-rtmp-cluster), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-rtmp-cluster))
* What is the process for building an RTMP Origin-Cluster? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-origin-cluster), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-origin-cluster))
* How can an HLS Edge-Cluster be set up?([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-hls-cluster), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-hls-cluster))
Here are some other important wikis:
* Usage: What is the method for delivering DASH (Experimental)? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-dash), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-dash))
* Usage: How can an RTMP stream be transcoded using FFMPEG? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-ffmpeg), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-ffmpeg))
* Usage: What is the process for setting up an HTTP FLV Live Streaming Cluster? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-http-flvCluster), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-http-flvCluster))
* Usage: How can HLS be delivered using an NGINX Cluster? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-hls-cluster), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-hls-cluster))
* Usage: What steps are to ingest a file, stream, or device to RTMP? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-ingest), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-ingest))
* Usage: How can a stream be forwarded to other servers? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/sample-forward), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/sample-forward))
* Usage: What are the strategies for improving edge performance on multiple CPUs? ([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/reuse-port), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/reuse-port))
* Usage: How can bugs be reported or contact be made with us? ([CN](https://ossrs.net/lts/zh-cn/contact), [EN](https://ossrs.io/lts/en-us/contact))
* [Winlin](https://github.com/winlinvip): Founder of the project, focusing on ST and Issues/PR. Responsible for architecture and maintenance.
* [ZhaoWenjie](https://github.com/wenjiegit): One of the earliest contributors, focusing on HDS and Windows. Has expertise in client technology.
* [ShiWei](https://github.com/runner365): Specializes in SRT and H.265, maintaining SRT and FLV patches for FFmpeg. An expert in codecs and FFmpeg.
* [XiaoZhihong](https://github.com/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](https://github.com/Bepartofyou): Focused on H.265, initially contributed to the FFmpeg module in SRS for transcoding AAC with OPUS for WebRTC.
* [XiaLixin](https://github.com/xialixin): Specializes in GB28181, with expertise in live streaming and WebRTC.
* [LiPeng](https://github.com/lipeng19811218): Concentrates on WebRTC and contributes to memory management and smart pointers.
* [ChenGuanghua](https://github.com/chen-guanghua): Focused on WebRTC/QoS and introduced the Asan toolchain to SRS.
* [ChenHaibo](https://github.com/duiniuluantanqin): Specializes in GB28181 and HTTP API, contributing to patches for FFmpeg with WHIP.
* [Genes](http://sourceforge.net/users/genes), [Mabbott](http://sourceforge.net/users/mabbott) and [Michael Talyanksy](https://github.com/michaeltalyansky) for creating and introducing [st](https://github.com/ossrs/state-threads/tree/srs).
