dotnet add package SIPSorcery -v 4.0.71-pre

Install-Package SIPSorcery -v 4.0.71-pre

dotnet new console --name SIPGetStarted -f netcoreapp3.1
cd SIPGetStarted
dotnet add package SIPSorcery -v 4.0.71-pre
dotnet add package SIPSorceryMedia.Windows -v 0.0.18-pre
code . # If you have Visual Studio Code https://code.visualstudio.com installed.
# edit Program.cs and paste in the contents below.
dotnet run
# if successful you will hear the current time read out.
ctrl-c

using System;
using System.Threading.Tasks;
using SIPSorcery.SIP.App;
using SIPSorcery.Media;
using SIPSorceryMedia.Windows;

namespace SIPGetStarted
{
    class Program
    {
         private static string DESTINATION = "time@sipsorcery.com";
        
        static async Task Main()
        {
            Console.WriteLine("SIP Get Started");
			
            var userAgent = new SIPUserAgent();
            var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
            var voipMediaSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());

            // Place the call and wait for the result.
            bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
            Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");

            Console.WriteLine("Press any key to hangup and exit.");
            Console.ReadLine();
        }
    }
}

dotnet new console --name WebRTCGetStarted -f netcoreapp3.1
cd WebRTCGetStarted
dotnet add package SIPSorcery -v 4.0.71-pre
dotnet add package SIPSorceryMedia.Windows -v 0.0.18-pre
dotnet add package Serilog.Sinks.Console
dotnet add package Serilog.Extensions.Logging
code . # If you have Visual Studio Code (https://code.visualstudio.com) installed
# edit Program.cs and paste in the contents below.
dotnet run

using System;
using System.Collections.Generic;
using System.Linq;
using System.Net;
using System.Threading;
using Microsoft.Extensions.Logging;
using Microsoft.Extensions.Logging.Abstractions;
using Serilog;
using SIPSorcery.Net;
using SIPSorceryMedia.Windows;
using WebSocketSharp.Server;
using SIPSorcery.Media;
using Serilog.Extensions.Logging;

namespace demo
{
    class Program
    {
        private const int WEBSOCKET_PORT = 8081;
        private const string STUN_URL = "stun:stun.sipsorcery.com";

        private static Microsoft.Extensions.Logging.ILogger logger = NullLogger.Instance;

        static void Main()
        {
            Console.WriteLine("WebRTC Get Started");

            logger = AddConsoleLogger();

            // Start web socket.
            Console.WriteLine("Starting web socket server...");
            var webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
            webSocketServer.AddWebSocketService<WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = CreatePeerConnection);
            webSocketServer.Start();

            Console.WriteLine($"Waiting for web socket connections on {webSocketServer.Address}:{webSocketServer.Port}...");
            Console.WriteLine("Press ctrl-c to exit.");

            // Ctrl-c will gracefully exit the call at any point.
            ManualResetEvent exitMre = new ManualResetEvent(false);
            Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
            {
                e.Cancel = true;
                exitMre.Set();
            };

            // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
            exitMre.WaitOne();
        }

        private static RTCPeerConnection CreatePeerConnection()
        {
            RTCConfiguration config = new RTCConfiguration
            {
                iceServers = new List<RTCIceServer> { new RTCIceServer { urls = STUN_URL } }
            };
            var pc = new RTCPeerConnection(config);

            var testPatternSource = new VideoTestPatternSource();
            WindowsVideoEndPoint windowsVideoEndPoint = new WindowsVideoEndPoint(true);
            var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });

            MediaStreamTrack videoTrack = new MediaStreamTrack(windowsVideoEndPoint.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
            pc.addTrack(videoTrack);
            MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
            pc.addTrack(audioTrack);

            testPatternSource.OnVideoSourceRawSample += windowsVideoEndPoint.ExternalVideoSourceRawSample;
            windowsVideoEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
            audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
            pc.OnVideoFormatsNegotiated += (sdpFormat) =>
                windowsVideoEndPoint.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
            pc.onconnectionstatechange += async (state) =>
            {
                logger.LogDebug($"Peer connection state change to {state}.");

                if (state == RTCPeerConnectionState.connected)
                {
                    await audioSource.StartAudio();
                    await windowsVideoEndPoint.StartVideo();
                    await testPatternSource.StartVideo();
                }
                else if (state == RTCPeerConnectionState.failed)
                {
                    pc.Close("ice disconnection");
                }
                else if (state == RTCPeerConnectionState.closed)
                {
                    await testPatternSource.CloseVideo();
                    await windowsVideoEndPoint.CloseVideo();
                    await audioSource.CloseAudio();
                }
            };

            // Diagnostics.
            pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
            pc.OnSendReport += (media, sr) => logger.LogDebug($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
            pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) => logger.LogDebug($"STUN {msg.Header.MessageType} received from {ep}.");
            pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");

            return pc;
        }

        /// <summary>
        ///  Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
        /// </summary>
        private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
        {
            var seriLogger = new LoggerConfiguration()
                .Enrich.FromLogContext()
                .MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
                .WriteTo.Console()
                .CreateLogger();
            var factory = new SerilogLoggerFactory(seriLogger);
            SIPSorcery.LogFactory.Set(factory);
            return factory.CreateLogger<Program>();
        }
    }
}

<!DOCTYPE html>
<head>
    <meta charset="UTF-8">

    <script type="text/javascript">

        const STUN_URL = "stun:stun.sipsorcery.com";
        const WEBSOCKET_URL = "ws://127.0.0.1:8081/"

        var pc, ws;

        async function start() {
            pc = new RTCPeerConnection({ iceServers: [{ urls: STUN_URL }] });

            pc.ontrack = evt => document.querySelector('#videoCtl').srcObject = evt.streams[0];
            pc.onicecandidate = evt => evt.candidate && ws.send(JSON.stringify(evt.candidate));
            await navigator.mediaDevices.getUserMedia({ video: true, audio: true })
                .then(stm => stm.getTracks().forEach(track => pc.addTrack(track, stm)));

            ws = new WebSocket(document.querySelector('#websockurl').value, []);
            ws.onmessage = async function (evt) {
                if (/^[\{"'\s]*candidate/.test(evt.data)) {
                    pc.addIceCandidate(JSON.parse(evt.data));
                }
                else {
                    await pc.setRemoteDescription(new RTCSessionDescription(JSON.parse(evt.data)));
                    pc.createAnswer()
                        .then((answer) => pc.setLocalDescription(answer))
                        .then(() => ws.send(JSON.stringify(pc.localDescription)));
                }
            };
        };

        async function closePeer() {
            pc.getSenders().forEach(sender => {
                sender.track.stop();
                pc.removeTrack(sender);
            });
            await pc.close();
            await ws.close();
        };

    </script>
</head>
<body>
    <video controls autoplay="autoplay" id="videoCtl" width="640" height="480"></video>
    <div>
        <input type="text" id="websockurl" size="40" />
        <button type="button" class="btn btn-success" onclick="start();">Start</button>
        <button type="button" class="btn btn-success" onclick="closePeer();">Close</button>
    </div>
</body>

<script>
    document.querySelector('#websockurl').value = WEBSOCKET_URL;
</script>

...
[19:40:25 DBG] STUN BindingRequest received from 192.168.0.50:57681.
[19:40:26 DBG] RTCP Receive for video from 192.168.11.50:57681
SDES: SSRC=3458092865, CNAME=5+ksoe4uBNfyl5u5
Sender: SSRC=3458092865, PKTS=18, BYTES=16392
[19:40:26 DBG] RTCP Receive for video from 192.168.11.50:57681
Receiver: SSRC=3458092865
 RR: SSRC=852075017, LOST=0, JITTER=390
[19:40:26 DBG] STUN BindingRequest received from 192.168.11.50:57681.
[19:40:27 DBG] RTCP Receive for video from 192.168.11.50:57681
SDES: SSRC=3458092865, CNAME=5+ksoe4uBNfyl5u5
Sender: SSRC=3458092865, PKTS=46, BYTES=39676
[19:40:27 DBG] RTCP Receive for video from 192.168.11.50:57681
Receiver: SSRC=3458092865
 RR: SSRC=852075017, LOST=0, JITTER=368
[19:40:27 DBG] STUN BindingRequest received from 192.168.11.50:57681.
[19:40:27 DBG] RTCP Receive for audio from 192.168.11.50:57681
SDES: SSRC=1049319500, CNAME=5+ksoe4uBNfyl5u5
Sender: SSRC=1049319500, PKTS=106, BYTES=16960
[19:40:27 DBG] STUN BindingRequest received from 192.168.0.50:57681.
[19:40:27 DBG] RTCP Receive for video from 192.168.11.50:57681
Receiver: SSRC=3458092865
 RR: SSRC=852075017, LOST=0, JITTER=419
 ...