SharpRTSPtoWebRTC 0.0.1-alpha

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 June 2023, H265 in WebRTC is only supported in Safari as an experimental feature. It has to be explicitly enabled by the user in Develop → Experimental Features → WebRTC H265 Codec. After enabling this option you should be able to play H265 video in the browser.

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:

"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:

builder.Services.AddSingleton<RTSPtoWebRTCProxyService>();

Then (optionally) add the configuration of streams from appsettings.json:

builder.Services.Configure<List<CameraConfiguration>>(builder.Configuration.GetSection("Cameras"));

Implement a minimal WebRTC signalling controller, for instance:

[ApiController]
[Route("api/[controller]")]
public class WebRTCController : ControllerBase
{
    private readonly IList<CameraConfiguration> _cameras;
    private readonly RTSPtoWebRTCProxyService _webRTCServer;

    public WebRTCController(IOptions<List<CameraConfiguration>> 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<IActionResult> 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.

Known issues

• A random crash due to a race condition in the sipsorcery 6.0.12 ICE implementation https://github.com/sipsorcery-org/sipsorcery/pull/924. Will be fixed when a new version of sipsorcery 6.0.13 is released.

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