diff --git a/.appveyor.yml b/.appveyor.yml
index 2bcefb4f5..f6be44588 100644
--- a/.appveyor.yml
+++ b/.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
diff --git a/examples/WebRTCExamples/WebRTCRenegotiateAudio/Program.cs b/examples/WebRTCExamples/WebRTCRenegotiateAudio/Program.cs
new file mode 100644
index 000000000..c181fbfcc
--- /dev/null
+++ b/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 (aaron@sipsorcery.com)
+// 
+// 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>();
+        }
+    }
+}
diff --git a/examples/WebRTCExamples/WebRTCRenegotiateAudio/WebRTCRenegotiateAudio.csproj b/examples/WebRTCExamples/WebRTCRenegotiateAudio/WebRTCRenegotiateAudio.csproj
new file mode 100644
index 000000000..df247ebdf
--- /dev/null
+++ b/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>
diff --git a/examples/WebRTCExamples/WebRTCRenegotiateAudio/webrtc.html b/examples/WebRTCExamples/WebRTCRenegotiateAudio/webrtc.html
new file mode 100644
index 000000000..d7b744a28
--- /dev/null
+++ b/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>
\ No newline at end of file