https://github.com/jimm98y/sharprtsptowebrtc
A C# implementation of the RTSP to WebRTC gateway that allows you to stream RTSP from various sources to the web browser. Supports H264 and H265 re-streaming. Audio transcoding from AAC to Opus is also supported.
https://github.com/jimm98y/sharprtsptowebrtc
aac h264 h265 rtsp streaming webrtc
Last synced: 6 months ago
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A C# implementation of the RTSP to WebRTC gateway that allows you to stream RTSP from various sources to the web browser. Supports H264 and H265 re-streaming. Audio transcoding from AAC to Opus is also supported.
- Host: GitHub
- URL: https://github.com/jimm98y/sharprtsptowebrtc
- Owner: jimm98y
- License: mit
- Created: 2023-04-07T13:37:40.000Z (over 2 years ago)
- Default Branch: main
- Last Pushed: 2025-04-05T13:55:03.000Z (6 months ago)
- Last Synced: 2025-04-05T14:02:41.421Z (6 months ago)
- Topics: aac, h264, h265, rtsp, streaming, webrtc
- Language: C#
- Homepage:
- Size: 3.19 MB
- Stars: 64
- Watchers: 5
- Forks: 13
- Open Issues: 3
-
Metadata Files:
- Readme: README.md
- License: LICENSE
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README
# SharpRTSP to WebRTC
This is a proof of concept bridge in between RTSP and WebRTC implemented in C#. It can take any H264/H265 RTSP stream and feed it through WebRTC to the web browser. It does not perform
any video transcoding which makes it lightweight and portable. It does support audio transcoding from AAC to Opus, all implemented in netstandard without any native dependencies.
## What can it do?
- Re-stream H264/H265 RTSP from any source to the web browser
- Stream aggregation - there is only a single session in between the gateway and the RTSP source, no matter how many users are watching the stream
- Transcode AAC audio to Opus with a small latency in audio
- Supports the experimental H265 WebRTC feature in Safari
## Compatibility
Because no video transcoding is being performed, the web browsers must support decoding of the source video codecs in WebRTC.
### H264
This should be supported by the majority of web browsers as it is among the codecs required by WebRTC. There might be an exception for Firefox on Android according to this: https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs.
### H265
Although most of the web browsers today support H265 video decoding, it does not mean H265 will also work in WebRTC. As of April 2025, H265 in WebRTC is supported in the latest releases of Safari. It is also supported in Chrome Canary 136+.
## Samples
### RTSPtoWebRTCGateway
There is a sample ASP.NET Core app that demonstrates the functionality on multiple live streams. To change the default configuration, just modify the `appsettings.json`:
```json
"Cameras": [
{
"Name": "name1",
"Url": "rtsp://url1",
"UserName": "MyUserName",
"Password": "MyPassword"
},
{
"Name": "name2",
"Url": "rtsp://url2",
"UserName": null,
"Password": null
}
]
```
### Minimal example
In `Startup.cs`, add the following piece of code to register the `RTSPtoWebRTCProxyService`:
```cs
builder.Services.AddSingleton();
```
Then (optionally) add the configuration of streams from `appsettings.json`:
```cs
builder.Services.Configure>(builder.Configuration.GetSection("Cameras"));
```
Implement a minimal WebRTC signalling controller, for instance:
```cs
[ApiController]
[Route("api/[controller]")]
public class WebRTCController : ControllerBase
{
private readonly IList _cameras;
private readonly RTSPtoWebRTCProxyService _webRTCServer;
public WebRTCController(IOptions> cameras, RTSPtoWebRTCProxyService webRTCServer)
{
_cameras = cameras.Value;
_webRTCServer = webRTCServer;
}
[HttpGet]
[Route("getcameras")]
public IActionResult GetCameras()
{
return Ok(_cameras.Select(x => x.Name).ToList());
}
[HttpGet]
[Route("getoffer")]
public async Task GetOffer(string id, string name)
{
return Ok(await _webRTCServer.GetOfferAsync(id, camera.Url, camera.UserName, camera.Password));
}
[HttpPost]
[Route("setanswer")]
public IActionResult SetAnswer(string id, [FromBody] RTCSessionDescriptionInit answer)
{
_webRTCServer.SetAnswer(id, answer);
return Ok();
}
[HttpPost]
[Route("addicecandidate")]
public IActionResult AddIceCandidate(string id, [FromBody] RTCIceCandidateInit iceCandidate)
{
_webRTCServer.AddIceCandidate(id, iceCandidate);
return Ok();
}
}
```
Finally, for the WebRTC viewer you can refer to `src/RTSPtoWebRTCGateway/ClientApp/src/components/CameraViewer.js`.
## Credits
- sipsorcery - WebRTC implementation in netstandard which has made this project possible https://github.com/sipsorcery-org/sipsorcery
- SharpRTSP - RTSP client in netstandard https://github.com/ngraziano/SharpRTSP
- concentus - Opus codec implementation https://github.com/lostromb/concentus
- SharpJaad.AAC - AAC decoder implementation https://github.com/jimm98y/SharpJaad
- Big thanks to AlexxIT for figuring out how the H265 WebRTC streaming in Safari works: https://github.com/AlexxIT/Blog/issues/5