/// <summary>
/// Runs a DTLS handshake test between two threads on a loopback address. The main motivation for
/// this test was that the first DTLS handshake between this application and a client browser
/// was often substantially slower and occasionally failed. By doing a loopback test the idea
/// is that the internal OpenSSL state is initialised.
/// </summary>
private static void DoDtlsHandshakeLoopbackTest()
{
IPAddress testAddr = IPAddress.Loopback;
//var dtlsFingerprint = DTLS_CERTIFICATE_FINGERPRINT.Substring(DTLS_CERTIFICATE_FINGERPRINT.Length + 1).Trim().Replace(":", "").Replace(" ", "");
//var dtlsFingerprintBuffer = SIPSorcery.Sys.ByteBufferInfo.ParseHexStr(dtlsFingerprint);
Socket svrSock = new Socket(testAddr.AddressFamily, SocketType.Dgram, ProtocolType.Udp);
svrSock.Bind(new IPEndPoint(testAddr, 9000));
int svrPort = ((IPEndPoint)svrSock.LocalEndPoint).Port;
DtlsHandshake svrHandshake = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
svrHandshake.Debug = true;
byte[] clientFingerprint = null;
var svrTask = Task.Run(() => svrHandshake.DoHandshakeAsServer((ulong)svrSock.Handle, ref clientFingerprint));
Socket cliSock = new Socket(testAddr.AddressFamily, SocketType.Dgram, ProtocolType.Udp);
cliSock.Bind(new IPEndPoint(testAddr, 0));
cliSock.Connect(testAddr, svrPort);
DtlsHandshake cliHandshake = new DtlsHandshake();
cliHandshake.Debug = true;
byte[] serverFingerprint = null;
var cliTask = Task.Run(() => cliHandshake.DoHandshakeAsClient((ulong)cliSock.Handle, (short)testAddr.AddressFamily, testAddr.GetAddressBytes(), (ushort)svrPort, ref serverFingerprint));
bool result = Task.WaitAll(new Task[] { svrTask, cliTask }, TEST_DTLS_HANDSHAKE_TIMEOUT);
cliHandshake.Shutdown();
svrHandshake.Shutdown();
cliSock.Close();
svrSock.Close();
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
/// <returns>True if the handshake completes successfully. False if not.</returns>
private static bool DoDtlsHandshake(WebRtcSession webRtcSession)
{
Console.WriteLine("DoDtlsHandshake started.");
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
webRtcSession.OnClose += (reason) => dtls.Shutdown();
int res = dtls.DoHandshakeAsServer((ulong)webRtcSession.RtpSession.RtpChannel.RtpSocket.Handle);
Console.WriteLine("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
Console.WriteLine("DTLS negotiation complete.");
// TODO fix race condition!!! First RTP packet is not getting decrypted.
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
webRtcSession.RtpSession.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
else
{
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
/// <returns>True if the handshake completed successfully or false otherwise.</returns>
private static bool DoDtlsHandshake(RTCPeerConnection peerConnection)
{
logger.LogDebug("DoDtlsHandshake started.");
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
dtls.Shutdown();
return(true);
}
else
{
dtls.Shutdown();
return(false);
}
}
/// <summary>
/// Runs a DTLS handshake test between two threads on a loopback address. The main motivation for
/// this test was that the first DTLS handshake between this application and a client browser
/// was often substantially slower and occasionally failed. By doing a loopback test the idea
/// is that the internal OpenSSL state is initialised.
/// </summary>
private void DoDtlsHandshakeLoopbackTest()
{
IPAddress testAddr = IPAddress.Loopback;
Socket svrSock = new Socket(testAddr.AddressFamily, SocketType.Dgram, ProtocolType.Udp);
svrSock.Bind(new IPEndPoint(testAddr, 9000));
int svrPort = ((IPEndPoint)svrSock.LocalEndPoint).Port;
DtlsHandshake svrHandshake = new DtlsHandshake(_dtlsCertificatePath, _dtlsKeyPath);
//svrHandshake.Debug = true;
var svrTask = Task.Run(() => svrHandshake.DoHandshakeAsServer((ulong)svrSock.Handle));
Socket cliSock = new Socket(testAddr.AddressFamily, SocketType.Dgram, ProtocolType.Udp);
cliSock.Bind(new IPEndPoint(testAddr, 0));
cliSock.Connect(testAddr, svrPort);
DtlsHandshake cliHandshake = new DtlsHandshake();
//cliHandshake.Debug = true;
var cliTask = Task.Run(() => cliHandshake.DoHandshakeAsClient((ulong)cliSock.Handle, (short)testAddr.AddressFamily, testAddr.GetAddressBytes(), (ushort)svrPort));
bool result = Task.WaitAll(new Task[] { svrTask, cliTask }, TEST_DTLS_HANDSHAKE_TIMEOUT);
cliHandshake.Shutdown();
svrHandshake.Shutdown();
cliSock.Close();
svrSock.Close();
}
static void Main()
{
Console.WriteLine("WebRTC Server Sample Program");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
ManualResetEvent exitMre = new ManualResetEvent(false);
AddConsoleLogger();
if (!File.Exists(MP4_FILE_PATH))
{
throw new ApplicationException($"The media file at does not exist at {MP4_FILE_PATH}.");
}
// Initialise OpenSSL & libsrtp, saves a couple of seconds for the first client connection.
Console.WriteLine("Initialising OpenSSL and libsrtp...");
DtlsHandshake.InitialiseOpenSSL();
Srtp.InitialiseLibSrtp();
Task.Run(DoDtlsHandshakeLoopbackTest).Wait();
Console.WriteLine("Test DTLS handshake complete.");
_mediaSource = new MediaSource();
_mediaSource.Init(MP4_FILE_PATH, true);
//_mediaSource.Init(0, 0, VideoSubTypesEnum.I420, 640, 480);
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2(WEBSOCKET_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
//_webSocketServer.Log.Level = WebSocketSharp.LogLevel.Debug;
_webSocketServer.AddWebSocketService <SDPExchange>("/", (sdpExchanger) =>
{
sdpExchanger.WebSocketOpened += SendSDPOffer;
sdpExchanger.SDPAnswerReceived += SDPAnswerReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Ctrl-c will gracefully exit the call at any point.
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();
_mediaSource.Shutdown();
_webSocketServer.Stop();
}
public Task StartAsync(CancellationToken cancellationToken)
{
_logger.LogDebug("WebRTCHostedService StartAsync.");
// Initialise OpenSSL & libsrtp, saves a couple of seconds for the first client connection.
_logger.LogDebug("Initialising OpenSSL and libsrtp...");
DtlsHandshake.InitialiseOpenSSL();
Srtp.InitialiseLibSrtp();
return(Task.CompletedTask);
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static bool DoDtlsHandshake(RTCPeerConnection peerConnection)
{
Log.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed)
{
dtls.Shutdown();
}
};
byte[] clientFingerprint = null;
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref clientFingerprint);
Log.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
// TODO: Check client fingerprint matches one supplied in the SDP.
Log.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
else
{
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
/// <returns>True if the handshake completed successfully or false otherwise.</returns>
private bool DoDtlsHandshake(WebRtcSession webRtcSession)
{
try
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(_dtlsCertificatePath))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {_dtlsCertificatePath}.");
}
else if (!File.Exists(_dtlsKeyPath))
{
throw new ApplicationException($"The DTLS key file could not be found at {_dtlsKeyPath}.");
}
var dtls = new DtlsHandshake(_dtlsCertificatePath, _dtlsKeyPath);
webRtcSession.OnClose += (reason) => dtls.Shutdown();
int res = dtls.DoHandshakeAsServer((ulong)webRtcSession.RtpSession.RtpChannel.RtpSocket.Handle);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
webRtcSession.RtpSession.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
webRtcSession.IsDtlsNegotiationComplete = true;
return(true);
}
else
{
return(false);
}
}
catch (Exception excp)
{
logger.LogWarning($"Exception DoDtlsHandshake. {excp}");
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static async Task <bool> DoDtlsHandshake(RTCPeerConnection peerConnection)
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
byte[] clientFingerprint = null;
var dtlsResult = await Task.Run(() => dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref clientFingerprint));
logger.LogDebug("DtlsContext initialisation result=" + dtlsResult);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS handshake succeeded.");
// TODO: Check client fingerprint matches one supplied in the SDP.
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
else
{
logger.LogWarning("DTLS handshake failed.");
dtls.Shutdown();
return(false);
}
}
static void Main()
{
Console.WriteLine("WebRTC Server Sample Program");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
ManualResetEvent exitMre = new ManualResetEvent(false);
AddConsoleLogger();
// Initialise OpenSSL & libsrtp, saves a couple of seconds for the first client connection.
Console.WriteLine("Initialising OpenSSL and libsrtp...");
DtlsHandshake.InitialiseOpenSSL();
Srtp.InitialiseLibSrtp();
InitialiseTestPattern();
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2(WEBSOCKET_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
//_webSocketServer.Log.Level = WebSocketSharp.LogLevel.Debug;
_webSocketServer.AddWebSocketService <SDPExchange>("/", (sdpExchanger) =>
{
sdpExchanger.WebSocketOpened += SendSDPOffer;
sdpExchanger.OnMessageReceived += WebSocketMessageReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
_sendTestPatternTimer?.Dispose();
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static int DoDtlsHandshake(WebRtcSession webRtcSession)
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
webRtcSession.OnClose += (reason) => dtls.Shutdown();
int res = dtls.DoHandshakeAsServer((ulong)webRtcSession.RtpSession.RtpChannel.RtpSocket.Handle);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
webRtcSession.RtpSession.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
if (!_isSampling)
{
Task.Run(StartMedia);
}
}
return(res);
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static bool DoDtlsHandshake(RTCPeerConnection peerConnection, DtlsHandshake dtls)
{
Log.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
byte[] clientFingerprint = null;
var dtlsResult = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref clientFingerprint);
Log.LogDebug($"DtlsContext initialisation result {dtlsResult}.");
if (dtls.IsHandshakeComplete())
{
Log.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
else
{
Log.LogWarning("DTLS handshake failed.");
dtls.Shutdown();
return(false);
}
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
}
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
peerConnection.addTrack(videoTrack);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) =>
{
peerConnection.Close("remote timeout");
dtls.Shutdown();
};
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
logger.LogDebug($"RTC peer connection was closed.");
OnMediaSampleReady -= peerConnection.SendMedia;
peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
peerConnection.OnSendReport -= RtpSession_OnSendReport;
}
else if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Peer connection connected.");
OnMediaSampleReady += peerConnection.SendMedia;
}
};
peerConnection.oniceconnectionstatechange += (state) =>
{
if (state == RTCIceConnectionState.connected)
{
logger.LogDebug("Starting DTLS handshake task.");
bool dtlsResult = false;
Task.Run(async() => dtlsResult = await DoDtlsHandshake(peerConnection, dtls, peerConnection.RemotePeerDtlsFingerprint))
.ContinueWith((t) =>
{
logger.LogDebug($"dtls handshake result {dtlsResult}.");
if (dtlsResult)
{
//peerConnection.SetDestination(SDPMediaTypesEnum.audio, peerConnection.IceSession.ConnectedRemoteEndPoint, peerConnection.IceSession.ConnectedRemoteEndPoint);
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
}
};
var offerInit = peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerInit);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
context.WebSocket.Send(offerInit.sdp);
return(peerConnection);
}
public async Task Start(CancellationToken ct)
{
try
{
AddConsoleLogger();
logger.LogDebug("WebRTCDaemon starting.");
var wssCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2(_webSocketCertificatePath, _webSocketCertificatePassword);
logger.LogDebug("Web Socket Server Certificate: " + wssCertificate.Subject + ", have key " + wssCertificate.HasPrivateKey + ", Expires " + wssCertificate.GetExpirationDateString() + ".");
logger.LogDebug($"DTLS certificate thumbprint {_dtlsCertificateThumbprint}.");
logger.LogDebug($"Web socket port {_webSocketPort}.");
if (!File.Exists(_mediaFilePath))
{
throw new ApplicationException($"The media file at does not exist at {_mediaFilePath}.");
}
// Initialise OpenSSL & libsrtp, saves a couple of seconds for the first client connection.
Console.WriteLine("Initialising OpenSSL and libsrtp...");
DtlsHandshake.InitialiseOpenSSL();
Srtp.InitialiseLibSrtp();
Task.Run(DoDtlsHandshakeLoopbackTest).Wait();
Console.WriteLine("Test DTLS handshake complete.");
// Configure the web socket and the different end point handlers.
int webSocketPort = (!String.IsNullOrEmpty(_webSocketPort)) ? Int32.Parse(_webSocketPort) : DEFAULT_WEB_SOCKET_PORT;
_webSocketServer = new WebSocketServer(IPAddress.Any, webSocketPort, true);
_webSocketServer.SslConfiguration.ServerCertificate = wssCertificate;
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
// Standard encrypted WebRtc stream.
_webSocketServer.AddWebSocketService <SDPExchange>("/max", (sdpReceiver) =>
{
sdpReceiver.MediaSource = MediaSourceEnum.Max;
sdpReceiver.WebSocketOpened += WebRtcStartCall;
sdpReceiver.SDPAnswerReceived += WebRtcAnswerReceived;
});
if (!String.IsNullOrEmpty(_testPatternImagePath) && File.Exists(_testPatternImagePath))
{
_webSocketServer.AddWebSocketService <SDPExchange>("/testpattern", (sdpReceiver) =>
{
sdpReceiver.MediaSource = MediaSourceEnum.TestPattern;
sdpReceiver.WebSocketOpened += WebRtcStartCall;
sdpReceiver.SDPAnswerReceived += WebRtcAnswerReceived;
});
}
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Initialise the Media Foundation library that will pull the samples from the mp4 file.
_mediaSource = new MediaSource();
_mediaSource.Init(_mediaFilePath, true);
_ = Task.Run(ExpireConnections);
while (!ct.IsCancellationRequested)
{
await Task.Delay(1000);
}
}
catch (Exception excp)
{
logger.LogError("Exception WebRTCDaemon.Start. " + excp);
}
finally
{
Stop();
}
}
static void Main()
{
Console.WriteLine("ICE Console Test Program");
Console.WriteLine("Press ctrl-c to exit.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
ManualResetEvent exitMre = new ManualResetEvent(false);
AddConsoleLogger();
// Initialise OpenSSL & libsrtp, saves a couple of seconds for the first client connection.
Console.WriteLine("Initialising OpenSSL and libsrtp...");
DtlsHandshake.InitialiseOpenSSL();
Srtp.InitialiseLibSrtp();
Task.Run(DoDtlsHandshakeLoopbackTest).Wait();
Console.WriteLine("Test DTLS handshake complete.");
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2(WEBSOCKET_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
//_webSocketServer.Log.Level = WebSocketSharp.LogLevel.Debug;
_webSocketServer.AddWebSocketService <WebRtcClient>("/sendoffer", (client) =>
{
client.WebSocketOpened += SendOffer;
client.OnMessageReceived += WebSocketMessageReceived;
});
_webSocketServer.AddWebSocketService <WebRtcClient>("/receiveoffer", (client) =>
{
client.WebSocketOpened += ReceiveOffer;
client.OnMessageReceived += WebSocketMessageReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Ctrl-c will gracefully exit the call at any point.
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();
_webSocketServer.Stop();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery sip.js Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var sipChannel = new SIPWebSocketChannel(IPAddress.Loopback, 80);
var wssCertificate = new System.Security.Cryptography.X509Certificates.X509Certificate2("localhost.pfx");
var sipChannelSecure = new SIPWebSocketChannel(IPAddress.Loopback, 443, wssCertificate);
sipTransport.AddSIPChannel(sipChannel);
sipTransport.AddSIPChannel(sipChannelSecure);
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogDebug($"Auto-answering incoming call from {req.Header.From}.");
var uas = userAgent.AcceptCall(req);
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
},
//X_RemoteSignallingAddress = context.UserEndPoint.Address,
//iceServers = new List<RTCIceServer> { new RTCIceServer { urls = SIPSORCERY_STUN_SERVER } }
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
peerConnection.OnTimeout += (mediaType) =>
{
peerConnection.Close("remote timeout");
};
peerConnection.oniceconnectionstatechange += async(state) =>
{
Log.LogDebug($"ICE connection state change to {state}.");
if (state == RTCIceConnectionState.connected)
{
var remoteEndPoint = peerConnection.AudioDestinationEndPoint;
Log.LogInformation($"ICE connected to remote end point {remoteEndPoint}.");
await Task.Run(() => DoDtlsHandshake(peerConnection, dtls))
.ContinueWith((dtlsResult) =>
{
Log.LogDebug($"dtls handshake result {dtlsResult.Result}.");
if (dtlsResult.Result)
{
var remoteEP = peerConnection.AudioDestinationEndPoint;
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
}
};
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.connected)
{
var remoteEP = peerConnection.AudioDestinationEndPoint;
Log.LogDebug($"DTLS connected on {remoteEP}.");
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
peerConnection.SetDestination(SDPMediaTypesEnum.video, remoteEP, remoteEP);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
// peerConnection.OnRtpPacketReceived += OnRtpPacketReceived;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendRecv);
peerConnection.addTrack(audioTrack);
//MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, MediaStreamStatusEnum.Inactive);
//peerConnection.addTrack(videoTrack);
var answerResult = await userAgent.Answer(uas, peerConnection);
};
Console.Write("press any key to exit...");
Console.Read();
sipTransport.Shutdown();
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static async Task <bool> DoDtlsHandshake(RTCPeerConnection peerConnection, DtlsHandshake dtls)
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
await peerConnection.Start();
//dtls.Shutdown();
if (_sendTestPatternTimer == null)
{
_sendTestPatternTimer = new Timer(SendTestPattern, null, 0, TEST_PATTERN_SPACING_MILLISECONDS);
}
return(true);
}
else
{
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
/// <returns>True if the handshake completed successfully or false otherwise.</returns>
private static async Task <bool> DoDtlsHandshake(RTCPeerConnection peerConnection, DtlsHandshake dtls, byte[] sdpFingerprint)
{
logger.LogDebug("DoDtlsHandshake started.");
byte[] clientFingerprint = null;
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref clientFingerprint);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
// TODO: The client fingerprint should be checked against the SDP fingerprint.
logger.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
await peerConnection.Start();
if (!_isSampling)
{
_ = Task.Run(StartMedia);
}
return(true);
}
else
{
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="peerConnection">The WebRTC session to perform the DTLS handshake on.</param>
/// <returns>True if the handshake completes successfully. False if not.</returns>
private static async Task <bool> DoDtlsHandshake(RTCPeerConnection peerConnection, DtlsHandshake dtls)
{
Console.WriteLine("DoDtlsHandshake started.");
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle);
Console.WriteLine("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
Console.WriteLine("DTLS negotiation complete.");
// TODO fix race condition!!! First RTP packet is not getting decrypted.
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
await peerConnection.Start();
return(true);
}
else
{
return(false);
}
}
private static async void WebSocketMessageReceived(WebSocketContext context, RTCPeerConnection peerConnection, string msg)
{
if (peerConnection.RemoteDescription != null)
{
Console.WriteLine($"ICE Candidate: {msg}.");
//await _peerConnections[0].addIceCandidate(new RTCIceCandidateInit { candidate = msg });
// await peerConnection.addIceCandidate(new RTCIceCandidateInit { candidate = msg });
Console.WriteLine("add ICE candidate complete.");
}
else
{
//Console.WriteLine($"websocket recv: {msg}");
//var offerSDP = SDP.ParseSDPDescription(msg);
Console.WriteLine($"offer sdp: {msg}");
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
await peerConnection.setRemoteDescription(new RTCSessionDescriptionInit { sdp = msg, type = RTCSdpType.offer });
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed)
{
Console.WriteLine($"webrtc session closed.");
//_peerConnections.Remove(peerConnection);
}
};
peerConnection.oniceconnectionstatechange += (state) =>
{
if (state == RTCIceConnectionState.connected)
{
Console.WriteLine("Starting DTLS handshake task.");
bool dtlsResult = false;
Task.Run(async() => dtlsResult = await DoDtlsHandshake(peerConnection, dtls))
.ContinueWith((t) =>
{
Console.WriteLine($"dtls handshake result {dtlsResult}.");
if (dtlsResult)
{
//peerConnection.SetDestination(SDPMediaTypesEnum.audio, peerConnection.IceSession.ConnectedRemoteEndPoint, peerConnection.IceSession.ConnectedRemoteEndPoint);
//_peerConnections.Add(peerConnection);
peerConnection.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
}
};
var answerInit = await peerConnection.createAnswer(null);
await peerConnection.setLocalDescription(answerInit);
Console.WriteLine($"answer sdp: {answerInit.sdp}");
context.WebSocket.Send(answerInit.sdp);
}
}
static async Task Main()
{
Console.WriteLine("WebRTC No Signalling Server Sample Program");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
ManualResetEvent exitMre = new ManualResetEvent(false);
AddConsoleLogger();
// Initialise OpenSSL & libsrtp, saves a couple of seconds for the first client connection.
Console.WriteLine("Initialising OpenSSL and libsrtp...");
DtlsHandshake.InitialiseOpenSSL();
Srtp.InitialiseLibSrtp();
InitialiseTestPattern();
Console.WriteLine($"Starting Peer Connection...");
var pc = await StartPeerConnection();
// Start the DTLS task before entering the ICE credentials to avoid DTLS client Hello retransmits.
// Start the DTLS task immediately so that we are ready to receive the client Hello.
// If it's missed it can take 10 or more seconds for the peer to retry.
var dtlsTask = DoDtlsHandshake(pc);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
_sendTestPatternTimer?.Dispose();
exitMre.Set();
};
Console.WriteLine("^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^");
Console.WriteLine("THE SDP OFFER ABOVE NEEDS TO BE PASTED INTO YOUR BROWSER");
Console.WriteLine();
Console.WriteLine("Press enter when you the SDP answer is available and then enter the ICE username and password...");
Console.ReadLine();
Console.Write("Enter the remote peer ICE User (e.g. for 'a=ice-ufrag:tGXy' enter tGXy) => ");
var remoteIceUser = Console.ReadLine();
Console.Write("Enter the remote peer ICE Password (e.g. for 'a=ice-pwd:Icew1/BpwUIJLn2dBMbQyYPB' enter Icew1/BpwUIJLn2dBMbQyYPB) => ");
var remoteIcePassword = Console.ReadLine();
pc.SetRemoteCredentials(remoteIceUser, remoteIcePassword);
var dtlsResult = await dtlsTask;
logger.LogDebug($"dtls handshake result {dtlsResult}.");
if (dtlsResult)
{
var remoteEP = pc.AudioDestinationEndPoint;
pc.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
else
{
pc.Close("dtls handshake failed.");
}
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
pc.Close("normal");
}
public async Task <RTCSessionDescriptionInit> GetOffer(string id)
{
if (string.IsNullOrWhiteSpace(id))
{
throw new ArgumentNullException("id", "A unique ID parameter must be supplied when creating a new peer connection.");
}
else if (_peerConnections.ContainsKey(id))
{
throw new ArgumentNullException("id", "The specified peer connection ID is already in use.");
}
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
}
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
MediaStreamTrack audioTrack = new MediaStreamTrack("0", SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
//peerConnection.OnRtpPacketReceived += (SDPMediaTypesEnum media, RTPPacket rtpPkt) => _logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}, SeqNum {rtpPkt.Header.SequenceNumber}.");
//peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
//peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) =>
{
peerConnection.Close("remote timeout");
dtls.Shutdown();
};
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
//peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
//peerConnection.OnSendReport -= RtpSession_OnSendReport;
_logger.LogDebug($"Peer connection {id} closed.");
_peerConnections.TryRemove(id, out _);
}
else if (state == RTCPeerConnectionState.connected)
{
_logger.LogDebug("Peer connection connected.");
}
};
_ = Task <bool> .Run(() => Task.FromResult(DoDtlsHandshake(peerConnection, dtls)))
.ContinueWith((t) =>
{
_logger.LogDebug($"dtls handshake result {t.Result}.");
if (t.Result)
{
var remoteEP = peerConnection.IceSession.ConnectedRemoteEndPoint;
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
var offerSdp = await peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerSdp);
_peerConnections.TryAdd(id, peerConnection);
return(offerSdp);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
}
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
MediaStreamTrack audioTrack = new MediaStreamTrack("0", SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) =>
{
peerConnection.Close("remote timeout");
dtls.Shutdown();
};
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
peerConnection.OnSendReport -= RtpSession_OnSendReport;
}
else if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Peer connection connected.");
peerConnection.OnRtpPacketReceived += (SDPMediaTypesEnum media, RTPPacket rtpPkt) => logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
}
};
peerConnection.oniceconnectionstatechange += (state) =>
{
if (state == RTCIceConnectionState.connected)
{
logger.LogDebug("Starting DTLS handshake task.");
bool dtlsResult = false;
Task.Run(async() => dtlsResult = await DoDtlsHandshake(peerConnection, dtls))
.ContinueWith((t) =>
{
logger.LogDebug($"dtls handshake result {dtlsResult}.");
if (dtlsResult)
{
var remoteEP = peerConnection.IceSession.ConnectedRemoteEndPoint;
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
}
};
var offerSdp = await peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
logger.LogDebug(offerSdp.sdp);
context.WebSocket.Send(offerSdp.sdp);
return(peerConnection);
}
