/// <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 Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
var testPatternSource = new VideoTestPatternSource();
var videoEncoderEndPoint = new VideoEncoderEndPoint();
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEncoderEndPoint.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
testPatternSource.OnVideoSourceRawSample += videoEncoderEndPoint.ExternalVideoSourceRawSample;
videoEncoderEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (formats) =>
videoEncoderEndPoint.SetVideoSourceFormat(formats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
await testPatternSource.StartVideo();
}
else if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await testPatternSource.CloseVideo();
await audioSource.CloseAudio();
}
};
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
/// <returns>True if the handshake completed successfully or false otherwise.</returns>
private static bool DoDtlsHandshake(RTCPeerConnection peerConnection)
{
logger.LogDebug("DoDtlsHandshake started.");
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
dtls.Shutdown();
return(true);
}
else
{
dtls.Shutdown();
return(false);
}
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
pc.createDataChannel("test", null);
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
};
// 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>
/// 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)
{
logger.LogDebug("DoDtlsHandshake started.");
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();
if (!_isSampling)
{
_ = Task.Run(StartMedia);
}
return(true);
}
else
{
return(false);
}
}
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);
MediaStreamTrack track = new MediaStreamTrack(windowsVideoEndPoint.GetVideoSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(track);
testPatternSource.OnVideoSourceRawSample += windowsVideoEndPoint.ExternalVideoSourceRawSample;
windowsVideoEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
pc.OnVideoFormatsNegotiated += (sdpFormat) =>
windowsVideoEndPoint.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await testPatternSource.CloseVideo();
await windowsVideoEndPoint.CloseVideo();
}
else if (state == RTCPeerConnectionState.connected)
{
await windowsVideoEndPoint.StartVideo();
await testPatternSource.StartVideo();
}
};
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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($"ICE connection state change to {state}.");
return(pc);
}
private static RTCPeerConnection Createpc(WebSocketContext context)
{
var pc = new RTCPeerConnection(null);
pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == STUNAttributeTypesEnum.UseCandidate);
Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
pc.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
//pc.OnReceiveReport += (type, rtcp) => logger.LogDebug($"RTCP {type} report received.");
pc.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
//pc.onicecandidate += (candidate) =>
//{
// if (pc.signalingState == RTCSignalingState.have_local_offer ||
// pc.signalingState == RTCSignalingState.have_remote_offer)
// {
// context.WebSocket.Send($"candidate:{candidate}");
// }
//};
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Creating RTP session for ffplay.");
var rtpSession = CreateRtpSession(pc.AudioLocalTrack?.Capabilities, pc.VideoLocalTrack?.Capabilities);
pc.OnRtpPacketReceived += (rep, media, rtpPkt) =>
{
if (media == SDPMediaTypesEnum.audio && rtpSession.AudioDestinationEndPoint != null)
{
//logger.LogDebug($"Forwarding {media} RTP packet to ffplay timestamp {rtpPkt.Header.Timestamp}.");
rtpSession.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
}
else if (media == SDPMediaTypesEnum.video && rtpSession.VideoDestinationEndPoint != null)
{
//logger.LogDebug($"Forwarding {media} RTP packet to ffplay timestamp {rtpPkt.Header.Timestamp}.");
rtpSession.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
}
};
pc.OnRtpClosed += (reason) => rtpSession.Close(reason);
}
};
_activePeerConnection = pc;
return(pc);
}
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;
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
pc.OnAudioFormatsNegotiated += (sdpFormat) =>
audioSource.SetAudioSourceFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static bool DoDtlsHandshake(RTCPeerConnection peerConnection)
{
Log.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed)
{
dtls.Shutdown();
}
};
byte[] clientFingerprint = null;
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref clientFingerprint);
Log.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
// TODO: Check client fingerprint matches one supplied in the SDP.
Log.LogDebug("DTLS negotiation complete.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
else
{
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static async Task <bool> DoDtlsHandshake(RTCPeerConnection peerConnection, DtlsHandshake dtls)
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
byte[] clientFingerprint = null;
int res = dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref clientFingerprint);
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
// TODO: Check client fingerprint matches one supplied in the SDP.
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
await peerConnection.Start();
//dtls.Shutdown();
if (_sendTestPatternTimer == null)
{
_sendTestPatternTimer = new Timer(SendTestPattern, null, 0, TEST_PATTERN_SPACING_MILLISECONDS);
}
return(true);
}
else
{
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static async Task <bool> DoDtlsHandshake(RTCPeerConnection peerConnection)
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
var dtls = new DtlsHandshake(DTLS_CERTIFICATE_PATH, DTLS_KEY_PATH);
var dtlsResult = await Task.Run(() => dtls.DoHandshakeAsServer((ulong)peerConnection.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle));
logger.LogDebug("DtlsContext initialisation result=" + dtlsResult);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS handshake succeeded.");
var srtpSendContext = new Srtp(dtls, false);
var srtpReceiveContext = new Srtp(dtls, true);
peerConnection.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
else
{
logger.LogWarning("DTLS handshake failed.");
dtls.Shutdown();
return(false);
}
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static bool DoDtlsHandshake(RTCPeerConnection peerConnection, DtlsHandshake dtls)
{
Log.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"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 Task <RTCPeerConnection> Createpc(WebSocketContext context, RTCIceServer stunServer, bool relayOnly)
{
if (_peerConnection != null)
{
_peerConnection.Close("normal");
}
List <RTCCertificate> presetCertificates = null;
if (File.Exists(LOCALHOST_CERTIFICATE_PATH))
{
var localhostCert = new X509Certificate2(LOCALHOST_CERTIFICATE_PATH, (string)null, X509KeyStorageFlags.Exportable);
presetCertificates = new List <RTCCertificate> {
new RTCCertificate {
Certificate = localhostCert
}
};
}
RTCConfiguration pcConfiguration = new RTCConfiguration
{
//certificates = presetCertificates,
//X_RemoteSignallingAddress = (context != null) ? context.UserEndPoint.Address : null,
iceServers = stunServer != null ? new List <RTCIceServer> {
stunServer
} : null,
iceTransportPolicy = relayOnly ? RTCIceTransportPolicy.relay : RTCIceTransportPolicy.all,
//X_BindAddress = IPAddress.Any, // NOTE: Not reqd. Using this to filter out IPv6 addresses so can test with Pion.
};
_peerConnection = new RTCPeerConnection(pcConfiguration);
//_peerConnection.GetRtpChannel().MdnsResolve = (hostname) => Task.FromResult(NetServices.InternetDefaultAddress);
_peerConnection.GetRtpChannel().MdnsResolve = MdnsResolve;
//_peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isrelay) => logger.LogDebug($"STUN message received from {ep}, message type {msg.Header.MessageType}.");
//var dc = _peerConnection.createDataChannel(DATA_CHANNEL_LABEL, null);
//dc.onmessage += (msg) => logger.LogDebug($"data channel receive ({dc.label}-{dc.id}): {msg}");
// Add a send-only audio track (this doesn't require any native libraries for encoding so is good for x-platform testing).
AudioExtrasSource audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
audioSource.OnAudioSourceEncodedSample += _peerConnection.SendAudio;
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
_peerConnection.addTrack(audioTrack);
_peerConnection.OnAudioFormatsNegotiated += (formats) =>
audioSource.SetAudioSourceFormat(formats.First());
_peerConnection.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
_peerConnection.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
_peerConnection.Close("remote disconnection");
}
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
}
else if (state == RTCPeerConnectionState.closed)
{
await audioSource.CloseAudio();
}
};
_peerConnection.OnReceiveReport += (re, media, rr) => logger.LogDebug($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
_peerConnection.OnSendReport += (media, sr) => logger.LogDebug($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
_peerConnection.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
// Peer ICE connection state changes are for ICE events such as the STUN checks completing.
_peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
_peerConnection.ondatachannel += (dc) =>
{
logger.LogDebug($"Data channel opened by remote peer, label {dc.label}, stream ID {dc.id}.");
dc.onmessage += (msg) =>
{
logger.LogDebug($"data channel ({dc.label}:{dc.id}): {msg}.");
};
};
_peerConnection.onsignalingstatechange += () =>
{
if (_peerConnection.signalingState == RTCSignalingState.have_remote_offer)
{
logger.LogDebug("Remote SDP:");
logger.LogDebug(_peerConnection.remoteDescription.sdp.ToString());
}
else if (_peerConnection.signalingState == RTCSignalingState.have_local_offer)
{
logger.LogDebug("Local SDP:");
logger.LogDebug(_peerConnection.localDescription.sdp.ToString());
}
};
return(Task.FromResult(_peerConnection));
}
private static RTCPeerConnection WebSocketOpened(WebSocketContext context)
{
WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint();
winVideoEP.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
_form.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = bmp)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap((int)width, (int)height, (int)(bmp.Length / height), System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_picBox.Image = bmpImage;
}
}
}));
};
var peerConnection = new RTCPeerConnection(null);
// Add local recvonly tracks. This ensures that the SDP answer includes only
// the codecs we support.
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(winVideoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == STUNAttributeTypesEnum.UseCandidate);
Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
peerConnection.oniceconnectionstatechange += (state) => Console.WriteLine($"ICE connection state changed to {state}.");
peerConnection.onconnectionstatechange += async(state) =>
{
Console.WriteLine($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.closed)
{
await winVideoEP.CloseVideo();
}
};
//peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
//{
// //logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
// if (media == SDPMediaTypesEnum.video)
// {
// winVideoEP.GotVideoRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
// }
//};
peerConnection.OnVideoFrameReceived += winVideoEP.GotVideoFrame;
return(peerConnection);
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var peerConnection = new RTCPeerConnection();
//FileStream captureStream = new FileStream("capture.stm", FileMode.Create, FileAccess.ReadWrite);
var videoEP = new Vp8NetVideoEncoderEndPoint();
videoEP.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = bmp)
{
Bitmap bmpImage = new Bitmap((int)width, (int)height, (int)(bmp.Length / height), PixelFormat.Format24bppRgb, (IntPtr)s);
_picBox.Image = bmpImage;
}
}
}));
}
};
// Sink (speaker) only audio end point.
//WindowsAudioEndPoint windowsAudioEP = new WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true, false);
//MediaStreamTrack audioTrack = new MediaStreamTrack(windowsAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.RecvOnly);
//peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnVideoFrameReceived += (rep, ts, frame, pixelFmt) =>
{
Console.WriteLine($"Video frame received {frame.Length} bytes.");
//Console.WriteLine(frame.HexStr());
//captureStream.Write(Encoding.ASCII.GetBytes($"{frame.Length},"));
//captureStream.Write(frame);
//captureStream.Flush();
videoEP.GotVideoFrame(rep, ts, frame, pixelFmt);
};
peerConnection.OnVideoFormatsNegotiated += (formats) =>
videoEP.SetVideoSinkFormat(formats.First());
//peerConnection.OnAudioFormatsNegotiated += (formats) =>
// windowsAudioEP.SetAudioSinkFormat(formats.First());
peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"Timeout on media {mediaType}.");
peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
peerConnection.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection connected changed to {state}.");
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.failed || state == RTCPeerConnectionState.disconnected)
{
//captureStream.Close();
}
};
peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == STUNAttributeTypesEnum.UseCandidate);
Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
//logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
//if (media == SDPMediaTypesEnum.audio)
//{
// windowsAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
//}
};
return(Task.FromResult(peerConnection));
}
private static RTCPeerConnection Createpc(WebSocketContext context, RTCIceServer stunServer)
{
if (_peerConnection != null)
{
_peerConnection.Close("normal");
}
List <RTCCertificate> presetCertificates = null;
if (File.Exists(LOCALHOST_CERTIFICATE_PATH))
{
var localhostCert = new X509Certificate2(LOCALHOST_CERTIFICATE_PATH, (string)null, X509KeyStorageFlags.Exportable);
presetCertificates = new List <RTCCertificate> {
new RTCCertificate {
Certificate = localhostCert
}
};
}
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = presetCertificates,
X_RemoteSignallingAddress = (context != null) ? context.UserEndPoint.Address : null,
iceServers = stunServer != null ? new List <RTCIceServer> {
stunServer
} : null,
iceTransportPolicy = RTCIceTransportPolicy.all,
//X_BindAddress = IPAddress.Any, // NOTE: Not reqd. Using this to filter out IPv6 addresses so can test with Pion.
};
_peerConnection = new RTCPeerConnection(pcConfiguration);
//_peerConnection.GetRtpChannel().MdnsResolve = (hostname) => Task.FromResult(NetServices.InternetDefaultAddress);
_peerConnection.GetRtpChannel().MdnsResolve = MdnsResolve;
_peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isrelay) => logger.LogDebug($"STUN message received from {ep}, message class {msg.Header.MessageClass}.");
var dc = _peerConnection.createDataChannel(DATA_CHANNEL_LABEL, null);
dc.onmessage += (msg) => logger.LogDebug($"data channel receive ({dc.label}-{dc.id}): {msg}");
// Add inactive audio and video tracks.
//MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.PCMU) }, MediaStreamStatusEnum.RecvOnly);
//pc.addTrack(audioTrack);
//MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, MediaStreamStatusEnum.Inactive);
//pc.addTrack(videoTrack);
_peerConnection.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
_peerConnection.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
_peerConnection.Close("remote disconnection");
}
};
_peerConnection.OnReceiveReport += (ep, type, rtcp) => logger.LogDebug($"RTCP {type} report received.");
_peerConnection.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
_peerConnection.onicecandidate += (candidate) =>
{
if (_peerConnection.signalingState == RTCSignalingState.have_local_offer ||
_peerConnection.signalingState == RTCSignalingState.have_remote_offer)
{
if (context != null)
{
context.WebSocket.Send($"candidate:{candidate}");
}
}
};
// Peer ICE connection state changes are for ICE events such as the STUN checks completing.
_peerConnection.oniceconnectionstatechange += (state) =>
{
logger.LogDebug($"ICE connection state change to {state}.");
};
_peerConnection.ondatachannel += (dc) =>
{
logger.LogDebug($"Data channel opened by remote peer, label {dc.label}, stream ID {dc.id}.");
dc.onmessage += (msg) =>
{
logger.LogDebug($"data channel ({dc.label}:{dc.id}): {msg}.");
};
};
return(_peerConnection);
}
/// <summary>
/// Hands the socket handle to the DTLS context and waits for the handshake to complete.
/// </summary>
/// <param name="webRtcSession">The WebRTC session to perform the DTLS handshake on.</param>
private static bool DoDtlsHandshake(RTCPeerConnection pc, SIPSorceryMedia.DtlsHandshake dtls, bool isClient, byte[] sdpFingerprint)
{
logger.LogDebug("DoDtlsHandshake started.");
if (!File.Exists(DTLS_CERTIFICATE_PATH))
{
throw new ApplicationException($"The DTLS certificate file could not be found at {DTLS_CERTIFICATE_PATH}.");
}
else if (!File.Exists(DTLS_KEY_PATH))
{
throw new ApplicationException($"The DTLS key file could not be found at {DTLS_KEY_PATH}.");
}
int res = 0;
bool fingerprintMatch = false;
if (isClient)
{
logger.LogDebug($"DTLS client handshake starting to {pc.AudioDestinationEndPoint}.");
// For the DTLS handshake to work connect must be called on the socket so openssl knows where to send.
var rtpSocket = pc.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket;
rtpSocket.Connect(pc.AudioDestinationEndPoint);
byte[] fingerprint = null;
var peerEP = pc.AudioDestinationEndPoint;
res = dtls.DoHandshakeAsClient((ulong)rtpSocket.Handle, (short)peerEP.AddressFamily.GetHashCode(), peerEP.Address.GetAddressBytes(), (ushort)peerEP.Port, ref fingerprint);
if (fingerprint != null)
{
logger.LogDebug($"DTLS server fingerprint {ByteBufferInfo.HexStr(fingerprint)}.");
fingerprintMatch = sdpFingerprint.SequenceEqual(fingerprint);
}
}
else
{
byte[] fingerprint = null;
res = dtls.DoHandshakeAsServer((ulong)pc.GetRtpChannel(SDPMediaTypesEnum.audio).RtpSocket.Handle, ref fingerprint);
if (fingerprint != null)
{
logger.LogDebug($"DTLS client fingerprint {ByteBufferInfo.HexStr(fingerprint)}.");
fingerprintMatch = sdpFingerprint.SequenceEqual(fingerprint);
}
}
logger.LogDebug("DtlsContext initialisation result=" + res);
if (dtls.IsHandshakeComplete())
{
logger.LogDebug("DTLS negotiation complete.");
if (!fingerprintMatch)
{
logger.LogWarning("DTLS fingerprint mismatch.");
return(false);
}
else
{
var srtpSendContext = new SIPSorceryMedia.Srtp(dtls, isClient);
var srtpReceiveContext = new SIPSorceryMedia.Srtp(dtls, !isClient);
pc.SetSecurityContext(
srtpSendContext.ProtectRTP,
srtpReceiveContext.UnprotectRTP,
srtpSendContext.ProtectRTCP,
srtpReceiveContext.UnprotectRTCP);
return(true);
}
}
else
{
return(false);
}
}
public async Task <RTCSessionDescriptionInit> GotOffer(RTCSessionDescriptionInit offer)
{
_logger.LogDebug($"SDP offer received.");
_logger.LogTrace($"Offer SDP:\n{offer.sdp}");
var pc = new RTCPeerConnection();
if (_publicIPv4 != null)
{
var rtpPort = pc.GetRtpChannel().RTPPort;
var publicIPv4Candidate = new RTCIceCandidate(RTCIceProtocol.udp, _publicIPv4, (ushort)rtpPort, RTCIceCandidateType.host);
pc.addLocalIceCandidate(publicIPv4Candidate);
}
if (_publicIPv6 != null)
{
var rtpPort = pc.GetRtpChannel().RTPPort;
var publicIPv6Candidate = new RTCIceCandidate(RTCIceProtocol.udp, _publicIPv6, (ushort)rtpPort, RTCIceCandidateType.host);
pc.addLocalIceCandidate(publicIPv6Candidate);
}
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(new VideoFormat(VideoCodecsEnum.VP8, VP8_PAYLOAD_ID), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
pc.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
//_logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}, SeqNum {rtpPkt.Header.SequenceNumber}.");
};
//peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
//peerConnection.OnSendReport += RtpSession_OnSendReport;
pc.OnTimeout += (mediaType) => _logger.LogWarning($"Timeout for {mediaType}.");
pc.onconnectionstatechange += (state) =>
{
_logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice failure");
}
};
var setResult = pc.setRemoteDescription(offer);
if (setResult == SetDescriptionResultEnum.OK)
{
var offerSdp = pc.createOffer(null);
await pc.setLocalDescription(offerSdp);
var answer = pc.createAnswer(null);
await pc.setLocalDescription(answer);
_logger.LogTrace($"Answer SDP:\n{answer.sdp}");
return(answer);
}
else
{
return(null);
}
}
public async Task <RTCSessionDescriptionInit> GotOffer(RTCSessionDescriptionInit offer)
{
logger.LogTrace($"SDP offer received.");
logger.LogTrace(offer.sdp);
var pc = new RTCPeerConnection();
if (_presetIceAddresses != null)
{
foreach (var addr in _presetIceAddresses)
{
var rtpPort = pc.GetRtpChannel().RTPPort;
var publicIPv4Candidate = new RTCIceCandidate(RTCIceProtocol.udp, addr, (ushort)rtpPort, RTCIceCandidateType.host);
pc.addLocalIceCandidate(publicIPv4Candidate);
}
}
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPWellKnownMediaFormatsEnum.PCMU);
pc.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(new VideoFormat(VideoCodecsEnum.VP8, VP8_PAYLOAD_ID));
pc.addTrack(videoTrack);
pc.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
};
pc.OnTimeout += (mediaType) => logger.LogWarning($"Timeout for {mediaType}.");
pc.oniceconnectionstatechange += (state) => logger.LogInformation($"ICE connection state changed to {state}.");
pc.onsignalingstatechange += () => logger.LogInformation($"Signaling state changed to {pc.signalingState}.");
pc.onconnectionstatechange += (state) =>
{
logger.LogInformation($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice failure");
}
};
var setResult = pc.setRemoteDescription(offer);
if (setResult == SetDescriptionResultEnum.OK)
{
var offerSdp = pc.createOffer(null);
await pc.setLocalDescription(offerSdp);
var answer = pc.createAnswer(null);
logger.LogTrace($"SDP answer created.");
logger.LogTrace(answer.sdp);
return(answer);
}
else
{
logger.LogWarning($"Failed to set remote description {setResult}.");
return(null);
}
}
private static RTCPeerConnection CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint();
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(false, 640, 480, 30);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(false, 1920, 1080, 30);
//await winVideoEP.InitialiseVideoSourceDevice();
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(winVideoEP.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
winVideoEP.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (sdpFormat) =>
winVideoEP.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
await winVideoEP.StartVideo();
}
else if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await winVideoEP.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);
}
private RTCPeerConnection Createpc()
{
List <RTCCertificate> presetCertificates = null;
if (File.Exists(LOCALHOST_CERTIFICATE_PATH))
{
var localhostCert = new X509Certificate2(LOCALHOST_CERTIFICATE_PATH, (string)null, X509KeyStorageFlags.Exportable);
presetCertificates = new List <RTCCertificate> {
new RTCCertificate {
Certificate = localhostCert
}
};
}
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = presetCertificates,
};
var pc = new RTCPeerConnection(pcConfiguration);
pc.GetRtpChannel().MdnsResolve = MdnsResolve;
//pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isrelay) => logger.LogDebug($"{_peerName}: STUN message received from {ep}, message class {msg.Header.MessageClass}.");
var dataChannel = pc.createDataChannel(_dataChannelLabel, null);
dataChannel.onDatamessage -= DataChannel_onDatamessage;
dataChannel.onDatamessage += DataChannel_onDatamessage;
_dataChannels.Add(_dataChannelLabel, dataChannel);
pc.onicecandidateerror += (candidate, error) => logger.LogWarning($"{_peerName}: Error adding remote ICE candidate. {error} {candidate}");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"{_peerName}: ICE connection state change to {state}.");
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"{_peerName}: Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
pc.Close("remote disconnection");
}
};
pc.onicecandidate += (candidate) =>
{
if (pc.signalingState == RTCSignalingState.have_local_offer ||
pc.signalingState == RTCSignalingState.have_remote_offer)
{
OnIceCandidateAvailable?.Invoke(new RTCIceCandidateInit()
{
candidate = candidate.ToString(),
sdpMid = candidate.sdpMid,
sdpMLineIndex = candidate.sdpMLineIndex
});
}
};
pc.ondatachannel += (dc) =>
{
dc.onopen += () => logger.LogDebug($"{_peerName}: Data channel now open label {dc.label}, stream ID {dc.id}.");
dc.onDatamessage -= DataChannel_onDatamessage;
dc.onDatamessage += DataChannel_onDatamessage;
logger.LogDebug($"{_peerName}: Data channel created by remote peer, label {dc.label}, stream ID {dc.id}.");
_dataChannels.Add(dc.label, dc);
};
return(pc);
}
private static async Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(new VpxVideoEncoder(), WEBCAM_NAME);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(new FFmpegVideoEncoder(), WEBCAM_NAME);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(WEBCAM_NAME, 1920, 1080, 30);
//winVideoEP.RestrictFormats(x => x.Codec == SIPSorceryMedia.Abstractions.V1.VideoCodecsEnum.H264);
bool initResult = await winVideoEP.InitialiseVideoSourceDevice();
if (!initResult)
{
throw new ApplicationException("Could not initialise video capture device.");
}
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(winVideoEP.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
winVideoEP.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (videoFormats) =>
winVideoEP.SetVideoSourceFormat(videoFormats.First());
pc.OnAudioFormatsNegotiated += (audioFormats) =>
audioSource.SetAudioSourceFormat(audioFormats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
await winVideoEP.StartVideo();
}
else if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await winVideoEP.CloseVideo();
await audioSource.CloseAudio();
}
};
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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($"ICE connection state change to {state}.");
return(pc);
}
private static RTCPeerConnection Createpc(WebSocketContext context)
{
var pc = new RTCPeerConnection(null);
pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == STUNAttributeTypesEnum.UseCandidate);
Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
pc.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
//pc.OnReceiveReport += (type, rtcp) => logger.LogDebug($"RTCP {type} report received.");
pc.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
//pc.onicecandidate += (candidate) =>
//{
// if (pc.signalingState == RTCSignalingState.have_local_offer ||
// pc.signalingState == RTCSignalingState.have_remote_offer)
// {
// context.WebSocket.Send($"candidate:{candidate}");
// }
//};
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Creating RTP session for ffplay.");
var rtpSession = CreateRtpSession(pc.AudioLocalTrack?.Capabilities, pc.VideoLocalTrack?.Capabilities);
Core.Initialize();
_libvlc = new LibVLC();
_mediaPlayer = new MediaPlayer(_libvlc);
//_mediaPlayer.Volume = 10;
var sdpFullPath = Path.Combine(Directory.GetParent(typeof(Program).Assembly.Location).FullName, FFPLAY_DEFAULT_SDP_PATH);
using var media = new Media(_libvlc, new Uri(sdpFullPath));
_mediaPlayer.Play(media);
//if (_activePeerConnection != null)
{
// request key frames
var localVideoSsrc = _activePeerConnection.VideoLocalTrack.Ssrc;
var remoteVideoSsrc = _activePeerConnection.VideoRemoteTrack.Ssrc;
RTCPFeedback pli = new RTCPFeedback(localVideoSsrc, remoteVideoSsrc, PSFBFeedbackTypesEnum.PLI);
_activePeerConnection.SendRtcpFeedback(SDPMediaTypesEnum.video, pli);
}
pc.OnRtpPacketReceived += (rep, media, rtpPkt) =>
{
if (media == SDPMediaTypesEnum.audio && rtpSession.AudioDestinationEndPoint != null)
{
//logger.LogDebug($"Forwarding {media} RTP packet to ffplay timestamp {rtpPkt.Header.Timestamp}.");
rtpSession.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
}
else if (media == SDPMediaTypesEnum.video && rtpSession.VideoDestinationEndPoint != null)
{
//logger.LogDebug($"Forwarding {media} RTP packet to ffplay timestamp {rtpPkt.Header.Timestamp}.");
rtpSession.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
}
};
pc.OnRtpClosed += (reason) => rtpSession.Close(reason);
}
};
_activePeerConnection = pc;
return(pc);
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Asterisk + ICE Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
userAgent.OnCallHungup += async(dialog) =>
{
// Give time for the BYE response before exiting.
await Task.Delay(1000);
exitCts.Cancel();
};
var audioExtras = new AudioExtrasSource();
audioExtras.SetSource(AudioSourcesEnum.PinkNoise);
var testPattern = new VideoTestPatternSource(new VpxVideoEncoder());
var pc = new RTCPeerConnection(null);
pc.OnAudioFormatsNegotiated += (formats) => audioExtras.SetAudioSourceFormat(formats.First());
pc.OnVideoFormatsNegotiated += (formats) => testPattern.SetVideoSourceFormat(formats.First());
var audioTrack = new MediaStreamTrack(audioExtras.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
audioExtras.OnAudioSourceEncodedSample += pc.SendAudio;
var videoTrack = new MediaStreamTrack(testPattern.GetVideoSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
testPattern.OnVideoSourceEncodedSample += pc.SendVideo;
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => Console.WriteLine($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
pc.OnSendReport += (media, sr) => Console.WriteLine($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) => Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}.");
pc.oniceconnectionstatechange += (state) => Console.WriteLine($"ICE connection state change to {state}.");
// ICE connection state handler.
pc.onconnectionstatechange += (state) =>
{
Console.WriteLine($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
audioExtras.StartAudio();
testPattern.StartVideo();
}
else if (state == RTCPeerConnectionState.failed)
{
if (userAgent.IsCallActive)
{
Console.WriteLine("ICE connection failed, hanging up active call.");
userAgent.Hangup();
}
}
};
// Place the call and wait for the result.
var callTask = userAgent.Call(DESTINATION, USERNAME, PASSWORD, pc);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent != null)
{
if (userAgent.IsCalling || userAgent.IsRinging)
{
Console.WriteLine("Cancelling in progress call.");
userAgent.Cancel();
}
else if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up established call.");
userAgent.Hangup();
}
}
;
exitCts.Cancel();
};
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
Console.WriteLine($"Call to {DESTINATION} succeeded.");
exitCts.Token.WaitHandle.WaitOne();
}
else
{
Console.WriteLine($"Call to {DESTINATION} failed.");
}
Console.WriteLine("Exiting...");
if (userAgent?.IsHangingUp == true)
{
Console.WriteLine("Waiting 1s for the call hangup or cancel to complete...");
await Task.Delay(1000);
}
// Clean up.
sipTransport.Shutdown();
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var peerConnection = new RTCPeerConnection(null);
var videoEP = new SIPSorceryMedia.Encoders.VideoEncoderEndPoint();
//var videoEP = new SIPSorceryMedia.Windows.WindowsEncoderEndPoint();
//var videoEP = new FFmpegVideoEndPoint();
videoEP.RestrictCodecs(new List <VideoCodecsEnum> {
VideoCodecsEnum.VP8
});
videoEP.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
_form.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = bmp)
{
Bitmap bmpImage = new Bitmap((int)width, (int)height, (int)(bmp.Length / height), PixelFormat.Format24bppRgb, (IntPtr)s);
_picBox.Image = bmpImage;
}
}
}));
};
// Sink (speaker) only audio end point.
WindowsAudioEndPoint windowsAudioEP = new WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(windowsAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnVideoFrameReceived += videoEP.GotVideoFrame;
peerConnection.OnVideoFormatsNegotiated += (sdpFormat) => videoEP.SetVideoSinkFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
peerConnection.OnAudioFormatsNegotiated += (sdpFormat) =>
windowsAudioEP.SetAudioSinkFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"Timeout on media {mediaType}.");
peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
peerConnection.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection connected changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await windowsAudioEP.StartAudio();
}
else if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.failed)
{
await windowsAudioEP.CloseAudio();
}
};
peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == STUNAttributeTypesEnum.UseCandidate);
Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
//logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
if (media == SDPMediaTypesEnum.audio)
{
windowsAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
}
};
return(Task.FromResult(peerConnection));
}
