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Added WebRTC demo that will use the VP8.Net encoder if/when it is ready. (#1359)
* Added WebRTC demo that will use the VP8.Net encoder if/when it is ready. * Remvoed some unused options.
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‎examples/SIPExamples/UserAgentClient/Program.cs‎

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using SIPSorcery.Net;
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using SIPSorcery.SIP;
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using SIPSorcery.SIP.App;
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using SIPSorceryMedia.Abstractions;
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using SIPSorceryMedia.Windows;
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namespace demo
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{
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class Program
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{
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//private static readonly string DEFAULT_DESTINATION_SIP_URI = "sips:[email protected]";
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private static readonly string DEFAULT_DESTINATION_SIP_URI = "sip:[email protected]";
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private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance;

‎examples/WebRTCExamples/WebRTCClientVP8Net/Program.cs‎

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using System;
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using System.Drawing;
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using System.Drawing.Imaging;
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using System.IO;
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using System.Linq;
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using System.Net;
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using System.Text;
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using System.Threading;
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using System.Threading.Tasks;
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using System.Windows.Forms;
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using Microsoft.Extensions.Logging;

‎examples/WebRTCExamples/WebRTCGetStartedVP8Net/Program.cs‎

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//-----------------------------------------------------------------------------
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// Filename: Program.cs
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//
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// Description: An example WebRTC server application that attempts to send a
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// test pattern to a browser peer and that uses a prototype .NET version of the
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// VP8 encoder. A web socket is used for signalling.
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//
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// The point of this demo is that it does not require any native libraries or
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// audio/video devices. This makes it a good palce to start for checking
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// whether a particular platform can be used to establish WebRTC connections
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// and get a media strem flowing.
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//
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// TODO: Not available until the VP8.NET project ports the encoder.
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//
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// Author(s):
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// Aaron Clauson ([email protected])
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//
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// History:
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// 12 Mar 2025 Aaron Clauson Created, Dublin, Ireland.
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//
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// License:
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// BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
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//-----------------------------------------------------------------------------
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using System;
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using System.Collections.Generic;
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using System.Linq;
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using System.Net;
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using System.Threading;
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using System.Threading.Tasks;
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using Microsoft.Extensions.Logging;
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using Microsoft.Extensions.Logging.Abstractions;
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using Serilog;
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using Serilog.Extensions.Logging;
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using SIPSorcery.Media;
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using SIPSorcery.Net;
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using Vpx.Net;
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using WebSocketSharp.Server;
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namespace demo
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{
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class Program
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{
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private const int WEBSOCKET_PORT = 8081;
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private const string STUN_URL = "stun:stun.cloudflare.com";
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private static Microsoft.Extensions.Logging.ILogger logger = NullLogger.Instance;
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static void Main()
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{
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Console.WriteLine("WebRTC Get Started");
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logger = AddConsoleLogger();
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// Start web socket.
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Console.WriteLine("Starting web socket server...");
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var webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
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webSocketServer.AddWebSocketService<WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = CreatePeerConnection);
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webSocketServer.Start();
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Console.WriteLine($"Waiting for web socket connections on {webSocketServer.Address}:{webSocketServer.Port}...");
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Console.WriteLine("Press ctrl-c to exit.");
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// Ctrl-c will gracefully exit the call at any point.
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ManualResetEvent exitMre = new ManualResetEvent(false);
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Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
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{
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e.Cancel = true;
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exitMre.Set();
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};
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// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
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exitMre.WaitOne();
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}
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private static Task<RTCPeerConnection> CreatePeerConnection()
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{
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RTCConfiguration config = new RTCConfiguration
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{
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iceServers = new List<RTCIceServer> { new RTCIceServer { urls = STUN_URL } },
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//X_BindAddress = IPAddress.Any
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};
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var pc = new RTCPeerConnection(config);
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var testPatternSource = new VideoTestPatternSource();
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var videoEncoderEndPoint = new Vp8NetVideoEncoderEndPoint();
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var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });
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MediaStreamTrack videoTrack = new MediaStreamTrack(videoEncoderEndPoint.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
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pc.addTrack(videoTrack);
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MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
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pc.addTrack(audioTrack);
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testPatternSource.OnVideoSourceRawSample += videoEncoderEndPoint.ExternalVideoSourceRawSample;
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videoEncoderEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
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audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
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pc.OnVideoFormatsNegotiated += (formats) => videoEncoderEndPoint.SetVideoSourceFormat(formats.First());
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pc.OnAudioFormatsNegotiated += (formats) => audioSource.SetAudioSourceFormat(formats.First());
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pc.onsignalingstatechange += () =>
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{
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logger.LogDebug($"Signalling state change to {pc.signalingState}.");
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if (pc.signalingState == RTCSignalingState.have_local_offer)
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{
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logger.LogDebug($"Local SDP offer:\n{pc.localDescription.sdp}");
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}
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else if (pc.signalingState == RTCSignalingState.stable)
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{
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logger.LogDebug($"Remote SDP offer:\n{pc.remoteDescription.sdp}");
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}
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};
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pc.onconnectionstatechange += async (state) =>
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{
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logger.LogDebug($"Peer connection state change to {state}.");
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if (state == RTCPeerConnectionState.connected)
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{
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await audioSource.StartAudio();
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await testPatternSource.StartVideo();
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}
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else if (state == RTCPeerConnectionState.failed)
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{
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pc.Close("ice disconnection");
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}
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else if (state == RTCPeerConnectionState.closed)
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{
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await testPatternSource.CloseVideo();
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await audioSource.CloseAudio();
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}
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};
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// Diagnostics.
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pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
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pc.OnSendReport += (media, sr) => logger.LogDebug($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
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pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) => logger.LogDebug($"STUN {msg.Header.MessageType} received from {ep}.");
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pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
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// To test closing.
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//_ = Task.Run(async () =>
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//{
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// await Task.Delay(5000);
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// audioSource.OnAudioSourceEncodedSample -= pc.SendAudio;
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// videoEncoderEndPoint.OnVideoSourceEncodedSample -= pc.SendVideo;
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// logger.LogDebug("Closing peer connection.");
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// pc.Close("normal");
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//});
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return Task.FromResult(pc);
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}
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/// <summary>
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/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
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/// </summary>
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private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
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{
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var seriLogger = new LoggerConfiguration()
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.Enrich.FromLogContext()
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.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
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.WriteTo.Console()
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.CreateLogger();
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var factory = new SerilogLoggerFactory(seriLogger);
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SIPSorcery.LogFactory.Set(factory);
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return factory.CreateLogger<Program>();
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}
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}
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}

‎examples/WebRTCExamples/WebRTCGetStartedVP8Net/WebRTCGetStartedVP8Net.csproj‎

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<Project Sdk="Microsoft.NET.Sdk">
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<PropertyGroup>
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<OutputType>Exe</OutputType>
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<TargetFramework>net8.0</TargetFramework>
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</PropertyGroup>
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<PropertyGroup>
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<AllowUnsafeBlocks>true</AllowUnsafeBlocks>
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<Prefer32Bit>false</Prefer32Bit>
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</PropertyGroup>
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<ItemGroup>
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<PackageReference Include="Microsoft.Extensions.Logging" Version="8.0.0" />
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<PackageReference Include="Serilog.Extensions.Logging" Version="3.0.1" />
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<PackageReference Include="Serilog.Sinks.Console" Version="3.1.1" />
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<PackageReference Include="SIPSorcery.WebSocketSharp" Version="0.0.1" />
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</ItemGroup>
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<ItemGroup>
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<ProjectReference Include="..\..\..\src\SIPSorcery.csproj" />
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<ProjectReference Include="..\..\..\..\VP8.Net\src\VP8.Net.csproj" />
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</ItemGroup>
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</Project>

‎examples/WebRTCExamples/WebRTCGetStartedVP8Net/getstarted.html‎

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<!DOCTYPE html>
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<head>
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<script type="text/javascript">
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const WEBSOCKET_URL = "ws://127.0.0.1:8081/"
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var pc, ws;
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async function start() {
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pc = new RTCPeerConnection();
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pc.ontrack = evt => {
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console.log("Adding track to video control.");
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document.querySelector('#videoCtl').srcObject = evt.streams[0];
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evt.streams[0].onunmute = () => {
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console.log("Adding track to video control.");
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};
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evt.streams[0].onended = () => {
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console.log("Track ended.");
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};
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}
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pc.onicecandidate = evt => evt.candidate && ws.send(JSON.stringify(evt.candidate));
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pc.onclose= () => {
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console.log("pc close");
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};
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ws = new WebSocket(document.querySelector('#websockurl').value, []);
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ws.onmessage = async function (evt) {
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var obj = JSON.parse(evt.data);
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if (obj?.candidate) {
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pc.addIceCandidate(obj);
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}
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else if (obj?.sdp) {
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await pc.setRemoteDescription(new RTCSessionDescription(obj));
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pc.createAnswer()
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.then((answer) => pc.setLocalDescription(answer))
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.then(() => ws.send(JSON.stringify(pc.localDescription)));
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}
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};
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};
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async function closePeer() {
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await pc?.close();
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await ws?.close();
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};
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</script>
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</head>
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<body>
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<video controls autoplay="autoplay" id="videoCtl" width="640" height="480"></video>
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<div>
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<input type="text" id="websockurl" size="40" />
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<button type="button" class="btn btn-success" onclick="start();">Start</button>
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<button type="button" class="btn btn-success" onclick="closePeer();">Close</button>
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</div>
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</body>
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<script>
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document.querySelector('#websockurl').value = WEBSOCKET_URL;
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</script>

‎examples/WebRTCExamples/WebRTCSendAudio/Program.cs‎

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//-----------------------------------------------------------------------------
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using System;
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using System.Collections.Generic;
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using System.Linq;
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using System.Net;
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using System.Threading;
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class Program
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{
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private const int WEBSOCKET_PORT = 8081;
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private const string STUN_URL = "stun:stun.sipsorcery.com";
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private static Microsoft.Extensions.Logging.ILogger logger = NullLogger.Instance;
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‎examples/WebRTCExamples/WebRTCSendAudio/WebRTCSendAudio.csproj‎

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<PackageReference Include="Serilog.Extensions.Logging" Version="3.0.1" />
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<PackageReference Include="Serilog.Sinks.Console" Version="3.1.1" />
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<PackageReference Include="SIPSorcery.WebSocketSharp" Version="0.0.1" />
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<PackageReference Include="SIPSorceryMedia.Windows" Version="8.0.7" />
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</ItemGroup>
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<ItemGroup>

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