Switch appveyor CI to VS Preview image for net5 builds.

.appveyor.yml

@@ -10,7 +10,7 @@ environment:
  access_token:
  # Token for github pages publishing.
  secure: 21m76jAVvcu7oACAHFnfCBltcwon+r5ZI3avfRmrNFAqJMn6RfLXwpBhPcJ617tD
-image: Visual Studio 2019
+image: Visual Studio 2019 Preview
 configuration: Release
 before_build:
 - cmd: nuget restore -DisableParallelProcessing src\SIPSorcery.sln

examples/WebRTCExamples/WebRTCRenegotiateAudio/Program.cs

@@ -0,0 +1,127 @@
+//-----------------------------------------------------------------------------
+// Filename: Program.cs
+//
+// Description: An example WebRTC server application that serves a sine wave 
+// audio stream to a WebRTC enabled browser.
+//
+// Author(s):
+// Aaron Clauson ([email protected])
+// 
+// History:
+// 28 Jul 2020 Aaron Clauson Created, Dublin, Ireland.
+//
+// License: 
+// BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file.
+//-----------------------------------------------------------------------------
+
+using System;
+using System.Collections.Generic;
+using System.Linq;
+using System.Net;
+using System.Threading;
+using System.Threading.Tasks;
+using Microsoft.Extensions.Logging;
+using Microsoft.Extensions.Logging.Abstractions;
+using Serilog;
+using Serilog.Extensions.Logging;
+using SIPSorcery.Net;
+using SIPSorcery.Media;
+using SIPSorceryMedia.Windows;
+using WebSocketSharp.Server;
+
+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 Audio Server Example Program");
+
+ 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($"Waiting for browser web socket connection to {webSocketServer.Address}:{webSocketServer.Port}...");
+
+ Console.WriteLine("Press ctrl-c to exit.");
+ ManualResetEvent exitMre = new ManualResetEvent(false);
+ Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
+ {
+ e.Cancel = true;
+ exitMre.Set();
+ };
+
+ // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
+ exitMre.WaitOne();
+ }
+
+ private static Task<RTCPeerConnection> CreatePeerConnection()
+ {
+ RTCConfiguration config = new RTCConfiguration
+ {
+ iceServers = new List<RTCIceServer> { new RTCIceServer { urls = STUN_URL } }
+ };
+ var pc = new RTCPeerConnection(config);
+
+ //AudioExtrasSource audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.SineWave });
+ //audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
+ WindowsAudioEndPoint audioSource = new WindowsAudioEndPoint(new AudioEncoder());
+ audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
+
+ MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
+ pc.addTrack(audioTrack);
+
+ pc.OnAudioFormatsNegotiated += (audioFormats) => audioSource.SetAudioSourceFormat(audioFormats.First());
+
+ pc.onconnectionstatechange += async (state) =>
+ {
+ logger.LogDebug($"Peer connection state change to {state}.");
+
+ if (state == RTCPeerConnectionState.connected)
+ {
+ await audioSource.StartAudio();
+ }
+ else if (state == RTCPeerConnectionState.failed)
+ {
+ pc.Close("ice disconnection");
+ }
+ else if (state == RTCPeerConnectionState.closed)
+ {
+ 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 Task.FromResult(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 serilogLogger = new LoggerConfiguration()
+ .Enrich.FromLogContext()
+ .MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
+ .WriteTo.Console()
+ .CreateLogger();
+ var factory = new SerilogLoggerFactory(serilogLogger);
+ SIPSorcery.LogFactory.Set(factory);
+ return factory.CreateLogger<Program>();
+ }
+ }
+}

examples/WebRTCExamples/WebRTCRenegotiateAudio/WebRTCRenegotiateAudio.csproj

@@ -0,0 +1,22 @@
+<Project Sdk="Microsoft.NET.Sdk">
+
+ <PropertyGroup>
+ <OutputType>Exe</OutputType>
+ <TargetFramework>netcoreapp3.1</TargetFramework>
+ </PropertyGroup>
+
+ <PropertyGroup>
+ <AllowUnsafeBlocks>true</AllowUnsafeBlocks>
+ </PropertyGroup>
+
+ <ItemGroup>
+ <PackageReference Include="Microsoft.Extensions.Logging" Version="3.1.8" />
+ <PackageReference Include="Microsoft.Extensions.Logging.Abstractions" Version="3.1.8" />
+ <PackageReference Include="Serilog.Extensions.Logging" Version="3.0.1" />
+ <PackageReference Include="Serilog.Sinks.Console" Version="3.1.1" />
+ <PackageReference Include="SIPSorcery" Version="4.0.88-pre" />
+ <PackageReference Include="SIPSorcery.WebSocketSharp" Version="0.0.1" />
+ <PackageReference Include="SIPSorceryMedia.Windows" Version="0.0.25-pre" />
+ </ItemGroup>
+
+</Project>

examples/WebRTCExamples/WebRTCRenegotiateAudio/webrtc.html

@@ -0,0 +1,59 @@
+<!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('#audioCtl').srcObject = evt.streams[0];
+ pc.onicecandidate = evt => evt.candidate && ws.send(JSON.stringify(evt.candidate));
+
+ // Diagnostics.
+ pc.onicegatheringstatechange = () => console.log("onicegatheringstatechange: " + pc.iceGatheringState);
+ pc.oniceconnectionstatechange = () => console.log("oniceconnectionstatechange: " + pc.iceConnectionState);
+ pc.onsignalingstatechange = () => console.log("onsignalingstatechange: " + pc.signalingState);
+ pc.onconnectionstatechange = () => console.log("onconnectionstatechange: " + pc.connectionState);
+
+ 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() {
+ await pc.close();
+ await ws.close();
+ };
+
+ </script>
+</head>
+<body>
+
+ <audio controls autoplay="autoplay" id="audioCtl"></audio>
+
+ <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>