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($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
//if (media == SDPMediaTypesEnum.audio)
//{
// windowsAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
//}
};
return(Task.FromResult(peerConnection));
}
public IMediaSource CreateMediaSource(MediaStreamTrack track, string id)
{
return(track?.CreateMediaSource());
}
static private Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
PeerConnection = new RTCPeerConnection(config);
switch (VideoSourceType)
{
case VIDEO_SOURCE.FILE_OR_STREAM:
// Do we use same file for Audio ?
if ((AudioSourceType == AUDIO_SOURCE.FILE_OR_STREAM) && (AudioSourceFile == VideoSourceFile))
{
SIPSorceryMedia.FFmpeg.FFmpegFileSource fileSource = new SIPSorceryMedia.FFmpeg.FFmpegFileSource(VideoSourceFile, RepeatVideoFile, new AudioEncoder(), true);
fileSource.OnEndOfFile += () => PeerConnection.Close("source eof");
videoSource = fileSource as IVideoSource;
audioSource = fileSource as IAudioSource;
}
else
{
SIPSorceryMedia.FFmpeg.FFmpegFileSource fileSource = new SIPSorceryMedia.FFmpeg.FFmpegFileSource(VideoSourceFile, RepeatVideoFile, new AudioEncoder(), true);
fileSource.OnEndOfFile += () => PeerConnection.Close("source eof");
videoSource = fileSource as IVideoSource;
}
break;
case VIDEO_SOURCE.CAMERA:
List <SIPSorceryMedia.FFmpeg.Camera>?cameras = SIPSorceryMedia.FFmpeg.FFmpegCameraManager.GetCameraDevices();
SIPSorceryMedia.FFmpeg.Camera?camera = null;
if (cameras?.Count > 0)
{
// Get last one
camera = cameras.Last();
}
if (camera != null)
{
videoSource = new SIPSorceryMedia.FFmpeg.FFmpegCameraSource(camera.Path);
}
else
{
throw new NotSupportedException($"Cannot find adequate camera ...");
}
break;
case VIDEO_SOURCE.SCREEN:
List <SIPSorceryMedia.FFmpeg.Monitor>?monitors = SIPSorceryMedia.FFmpeg.FFmpegMonitorManager.GetMonitorDevices();
SIPSorceryMedia.FFmpeg.Monitor? primaryMonitor = null;
if (monitors?.Count > 0)
{
foreach (SIPSorceryMedia.FFmpeg.Monitor monitor in monitors)
{
if (monitor.Primary)
{
primaryMonitor = monitor;
break;
}
}
if (primaryMonitor == null)
{
primaryMonitor = monitors[0];
}
}
if (primaryMonitor != null)
{
videoSource = new SIPSorceryMedia.FFmpeg.FFmpegScreenSource(primaryMonitor.Path, primaryMonitor.Rect, 10);
}
else
{
throw new NotSupportedException($"Cannot find adequate monitor ...");
}
break;
}
if (audioSource == null)
{
switch (AudioSourceType)
{
case AUDIO_SOURCE.FILE_OR_STREAM:
SIPSorceryMedia.FFmpeg.FFmpegFileSource fileSource = new SIPSorceryMedia.FFmpeg.FFmpegFileSource(AudioSourceFile, RepeatAudioFile, new AudioEncoder(), false);
fileSource.OnEndOfFile += () => PeerConnection.Close("source eof");
audioSource = fileSource as IAudioSource;
break;
case AUDIO_SOURCE.MICROPHONE:
audioSource = new SIPSorceryMedia.FFmpeg.FFmpegMicrophoneSource(MicrophoneDevicePath, new AudioEncoder());
break;
}
}
if (videoSource != null)
{
videoSource.RestrictFormats(x => x.Codec == VideoCodec);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoSource.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
PeerConnection.addTrack(videoTrack);
videoSource.OnVideoSourceEncodedSample += PeerConnection.SendVideo;
PeerConnection.OnVideoFormatsNegotiated += (videoFormats) => videoSource.SetVideoSourceFormat(videoFormats.First());
}
if (audioSource != null)
{
audioSource.RestrictFormats(x => x.Codec == AudioCodec);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
PeerConnection.addTrack(audioTrack);
audioSource.OnAudioSourceEncodedSample += AudioSource_OnAudioSourceEncodedSample;
PeerConnection.OnAudioFormatsNegotiated += (audioFormats) => audioSource.SetAudioSourceFormat(audioFormats.First());
}
PeerConnection.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
PeerConnection.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
if (videoSource != null)
{
await videoSource.CloseVideo();
}
if (audioSink != null)
{
await audioSink.CloseAudioSink();
}
if (audioSource != null)
{
await audioSource.CloseAudio();
}
}
else if (state == RTCPeerConnectionState.connected)
{
if (videoSource != null)
{
await videoSource.StartVideo();
}
if (audioSink != null)
{
await audioSink.StartAudioSink();
}
if (audioSource != null)
{
await audioSource.StartAudio();
}
}
};
// 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($"STUN {msg.Header.MessageType} received from {ep}.");
PeerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
return(Task.FromResult(PeerConnection));
}
public virtual RTCRtpTransceiver AddTransceiver(string connectionId, MediaStreamTrack track, RTCRtpTransceiverDirection direction = RTCRtpTransceiverDirection.SendRecv)
{
return(m_handler.AddTransceiver(connectionId, track, direction));
}
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.RestrictFormats(format => format.Codec == 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 += (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 += 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($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
if (media == SDPMediaTypesEnum.audio)
{
windowsAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
}
};
return(Task.FromResult(peerConnection));
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var videoEP = new SIPSorceryMedia.FFmpeg.FFmpegVideoEndPoint();
videoEP.RestrictFormats(format => format.Codec == VideoCodecsEnum.H264);
videoEP.OnVideoSinkDecodedSampleFaster += (RawImage rawImage) =>
{
_form.BeginInvoke(new Action(() =>
{
unsafe
{
if (_picBox.Width != rawImage.Width || _picBox.Height != rawImage.Height)
{
logger.LogDebug($"Adjusting video display from {_picBox.Width}x{_picBox.Height} to {rawImage.Width}x{rawImage.Height}.");
_picBox.Width = rawImage.Width;
_picBox.Height = rawImage.Height;
}
Bitmap bmpImage = new Bitmap(rawImage.Width, rawImage.Height, rawImage.Stride, PixelFormat.Format24bppRgb, rawImage.Sample);
_picBox.Image = bmpImage;
}
}));
};
RTCConfiguration config = new RTCConfiguration
{
//iceServers = new List<RTCIceServer> { new RTCIceServer { urls = STUN_URL } }
X_UseRtpFeedbackProfile = true
};
var pc = new RTCPeerConnection(config);
// Add local receive only tracks. This ensures that the SDP answer includes only the codecs we support.
if (!_options.NoAudio)
{
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
pc.addTrack(audioTrack);
pc.OnRtpPacketReceived += Pc_OnRtpPacketReceived;
}
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
//MediaStreamTrack videoTrack = new MediaStreamTrack(new VideoFormat(96, "VP8", 90000, "x-google-max-bitrate=5000000"), MediaStreamStatusEnum.RecvOnly);
pc.addTrack(videoTrack);
pc.OnVideoFrameReceived += videoEP.GotVideoFrame;
pc.OnVideoFormatsNegotiated += (formats) => videoEP.SetVideoSinkFormat(formats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await videoEP.CloseVideo();
}
else if (state == RTCPeerConnectionState.connected)
{
await videoEP.StartVideoSink();
}
};
// Diagnostics.
//pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
pc.OnSendReport += (media, sr) => logger.LogDebug($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
//pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) => logger.LogDebug($"RECV STUN {msg.Header.MessageType} (txid: {msg.Header.TransactionId.HexStr()}) from {ep}.");
//pc.GetRtpChannel().OnStunMessageSent += (msg, ep, isRelay) => logger.LogDebug($"SEND STUN {msg.Header.MessageType} (txid: {msg.Header.TransactionId.HexStr()}) to {ep}.");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
public async Task HandleInvalidSdpPortOnPlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// This transport will act as the call receiver. It allows the test to provide a
// tailored response to an incoming call.
SIPTransport calleeTransport = new SIPTransport();
// This transport will be used by the SIPUserAgent being tested to place the call.
SIPTransport callerTransport = new SIPTransport();
RTPSession rtpSession = new RTPSession(false, false, false);
try
{
calleeTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
calleeTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
if (req.Method != SIPMethodsEnum.INVITE)
{
SIPResponse notAllowedResponse = SIPResponse.GetResponse(req, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await calleeTransport.SendResponseAsync(notAllowedResponse);
}
else
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(calleeTransport, req, null);
var uas = new SIPServerUserAgent(calleeTransport, null, null, null, SIPCallDirection.In, null, null, uasTransaction);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = @"
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp, null, SIPDialogueTransferModesEnum.NotAllowed);
}
};
SIPUserAgent userAgent = new SIPUserAgent(callerTransport, null);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
SIPURI dstUri = new SIPURI(SIPSchemesEnum.sip, calleeTransport.GetSIPChannels().First().ListeningSIPEndPoint);
var result = await userAgent.Call(dstUri.ToString(), null, null, rtpSession);
Assert.False(result);
}
finally
{
rtpSession?.Close("normal");
callerTransport?.Shutdown();
calleeTransport?.Shutdown();
}
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var videoEP = new SIPSorceryMedia.Windows.WindowsVideoEndPoint();
//var videoEP = new SIPSorceryMedia.FFmpeg.FFmpegVideoEndPoint();
videoEP.RestrictFormats(format => format.Codec == 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;
}
}
}));
};
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
// Add local receive only tracks. This ensures that the SDP answer includes only the codecs we support.
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
pc.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
//MediaStreamTrack videoTrack = new MediaStreamTrack(new VideoFormat(96, "VP8", 90000, "x-google-max-bitrate=5000000"), MediaStreamStatusEnum.RecvOnly);
pc.addTrack(videoTrack);
pc.OnVideoFrameReceived += videoEP.GotVideoFrame;
pc.OnVideoFormatsNegotiated += (formats) => videoEP.SetVideoSinkFormat(formats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await videoEP.CloseVideo();
}
};
// Diagnostics.
//pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
public void CheckNoAudioNegotiationUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// By default offers made by us always put audio first. Create a remote SDP offer
// with the video first.
string remoteSdp =
@"v=0
o=- 62533 0 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS
m=audio 57148 UDP/TLS/RTP/SAVP 0 101
c=IN IP6 2a02:8084:6981:7880::76
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2944 1 udp 659136 192.168.11.50 57148 typ host generation 0
a=candidate:2488 1 udp 659136 192.168.0.50 57148 typ host generation 0
a=candidate:2507 1 udp 659136 fe80::54a9:d238:b2ee:ceb%24 57148 typ host generation 0
a=candidate:3159 1 udp 659136 2a02:8084:6981:7880::76 57148 typ host generation 0
a=ice-ufrag:CUTK
a=ice-pwd:QTCZWDIEBCIBGOYAGSIXRFIL
a=ice-options:ice2,trickle
a=fingerprint:sha-256 06:2F:61:85:1F:83:64:88:1B:93:93:8C:E5:FF:1C:D9:82:EA:60:97:1E:0D:DA:FA:28:11:00:FA:74:69:23:DB
a=setup:actpass
a=mid:0
a=sendrecv
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 9 UDP/TLS/RTP/SAVP 100
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:CUTK
a=ice-pwd:QTCZWDIEBCIBGOYAGSIXRFIL
a=ice-options:ice2,trickle
a=fingerprint:sha-256 06:2F:61:85:1F:83:64:88:1B:93:93:8C:E5:FF:1C:D9:82:EA:60:97:1E:0D:DA:FA:28:11:00:FA:74:69:23:DB
a=setup:actpass
a=mid:1
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000";
// Create a local session and add the video track first.
RTCPeerConnection pc = new RTCPeerConnection(null);
MediaStreamTrack localVideoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.video, 96, "VP8", 90000)
});
pc.addTrack(localVideoTrack);
var offer = SDP.ParseSDPDescription(remoteSdp);
logger.LogDebug($"Remote offer: {offer}");
var result = pc.SetRemoteDescription(SIP.App.SdpType.offer, offer);
logger.LogDebug($"Set remote description on local session result {result}.");
Assert.Equal(SetDescriptionResultEnum.OK, result);
var answer = pc.CreateAnswer(null);
logger.LogDebug($"Local answer: {answer}");
Assert.Equal(MediaStreamStatusEnum.Inactive, pc.AudioLocalTrack.StreamStatus);
Assert.Equal(100, pc.VideoLocalTrack.Capabilities.Single(x => x.Name() == "VP8").ID);
pc.Close("normal");
}
public void AddTrack(MediaStreamTrack track)
{
tracks.Add(track);
}
public MediaStreamTrackNative(MediaStreamTrack mediaStreamTrack) : base(mediaStreamTrack)
{
_mediaStreamTrack = mediaStreamTrack;
}
private static async Task <RTCPeerConnection> StartPeerConnection()
{
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);
MediaStreamTrack videoTrack = new MediaStreamTrack(
"0",
SDPMediaTypesEnum.video,
false,
new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
},
MediaStreamStatusEnum.SendOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
//peerConnection.OnRtcpBye += (reason) =>
//{
// logger.LogInformation("RTCP BYE report received from remote peer.");
// peerConnection.Close(reason);
// dtls.Shutdown();
//};
peerConnection.OnTimeout += (mediaType) =>
{
peerConnection.Close("remote timeout");
};
peerConnection.onconnectionstatechange += async(state) =>
{
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
OnTestPatternSampleReady -= peerConnection.SendMedia;
peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
peerConnection.OnSendReport -= RtpSession_OnSendReport;
}
else if (state == RTCPeerConnectionState.connected)
{
// The DTLS handshake completed.
logger.LogDebug("Peer connection connected.");
OnTestPatternSampleReady += peerConnection.SendMedia;
await peerConnection.Start();
if (_sendTestPatternTimer == null)
{
_sendTestPatternTimer = new Timer(SendTestPattern, null, 0, TEST_PATTERN_SPACING_MILLISECONDS);
}
}
};
peerConnection.oniceconnectionstatechange += (state) =>
{
logger.LogDebug($"ICE connection state change {state}.");
// The ICE connectivity check completed successfully.
if (state == RTCIceConnectionState.connected)
{
var remoteEP = peerConnection.IceSession.ConnectedRemoteEndPoint;
peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
};
var offerSdp = await peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerSdp);
logger.LogDebug(offerSdp.sdp);
return(peerConnection);
}
private static Task <RTCPeerConnection> Createpc(WebSocketContext context, RTCIceServer stunServer, bool relayOnly)
{
if (_peerConnection != null)
{
_peerConnection.Close("normal");
}
List <RTCCertificate> presetCertificates = null;
if (File.Exists(LOCALHOST_CERTIFICATE_PATH))
{
var localhostCert = new X509Certificate2(LOCALHOST_CERTIFICATE_PATH, (string)null, X509KeyStorageFlags.Exportable);
presetCertificates = new List <RTCCertificate> {
new RTCCertificate {
Certificate = localhostCert
}
};
}
RTCConfiguration pcConfiguration = new RTCConfiguration
{
//certificates = presetCertificates,
//X_RemoteSignallingAddress = (context != null) ? context.UserEndPoint.Address : null,
iceServers = stunServer != null ? new List <RTCIceServer> {
stunServer
} : null,
iceTransportPolicy = relayOnly ? RTCIceTransportPolicy.relay : RTCIceTransportPolicy.all,
//X_BindAddress = IPAddress.Any, // NOTE: Not reqd. Using this to filter out IPv6 addresses so can test with Pion.
};
_peerConnection = new RTCPeerConnection(pcConfiguration);
//_peerConnection.GetRtpChannel().MdnsResolve = (hostname) => Task.FromResult(NetServices.InternetDefaultAddress);
_peerConnection.GetRtpChannel().MdnsResolve = MdnsResolve;
//_peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isrelay) => logger.LogDebug($"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($"data channel ({dc.label}:{dc.id}): {msg}.");
};
};
return(Task.FromResult(_peerConnection));
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"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);
}
/// <summary>
///
/// </summary>
/// <param name="connectionId"></param>
/// <param name="track"></param>
public void RemoveSenderTrack(string connectionId, MediaStreamTrack track)
{
var sender = GetSenders(connectionId).First(s => s.Track == track);
_mapConnectionIdAndPeer[connectionId].peer.RemoveTrack(sender);
}
private static Task <RTCPeerConnection> CreatePeerConnection(X509Certificate2 cert)
{
//RTCConfiguration config = new RTCConfiguration
//{
// iceServers = new List<RTCIceServer> { new RTCIceServer { urls = STUN_URL } },
// certificates = new List<RTCCertificate> { new RTCCertificate { Certificate = cert } }
//};
//var pc = new RTCPeerConnection(config);
var pc = new RTCPeerConnection(null);
//var testPatternSource = new VideoTestPatternSource(new SIPSorceryMedia.Encoders.VideoEncoder());
var testPatternSource = new VideoTestPatternSource(new FFmpegVideoEncoder());
testPatternSource.SetFrameRate(TEST_PATTERN_FRAMES_PER_SECOND);
//testPatternSource.SetMaxFrameRate(true);
//var videoEndPoint = new SIPSorceryMedia.FFmpeg.FFmpegVideoEndPoint();
//videoEndPoint.RestrictFormats(format => format.Codec == VideoCodecsEnum.H264);
//testPatternSource.RestrictFormats(format => format.Codec == VideoCodecsEnum.H264);
//var videoEndPoint = new SIPSorceryMedia.Windows.WindowsEncoderEndPoint();
//var videoEndPoint = new SIPSorceryMedia.Encoders.VideoEncoderEndPoint();
MediaStreamTrack track = new MediaStreamTrack(testPatternSource.GetVideoSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(track);
//testPatternSource.OnVideoSourceRawSample += videoEndPoint.ExternalVideoSourceRawSample;
testPatternSource.OnVideoSourceRawSample += MesasureTestPatternSourceFrameRate;
testPatternSource.OnVideoSourceEncodedSample += pc.SendVideo;
pc.OnVideoFormatsNegotiated += (formats) => testPatternSource.SetVideoSourceFormat(formats.First());
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();
testPatternSource.Dispose();
}
else if (state == RTCPeerConnectionState.connected)
{
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}.");
pc.onsignalingstatechange += () =>
{
if (pc.signalingState == RTCSignalingState.have_local_offer)
{
logger.LogDebug($"Local SDP set, type {pc.localDescription.type}.");
logger.LogDebug(pc.localDescription.sdp.ToString());
}
else if (pc.signalingState == RTCSignalingState.have_remote_offer)
{
logger.LogDebug($"Remote SDP set, type {pc.remoteDescription.type}.");
logger.LogDebug(pc.remoteDescription.sdp.ToString());
}
};
return(Task.FromResult(pc));
}
/// <summary>
///
/// </summary>
/// <param name="connectionId"></param>
/// <param name="track"></param>
/// <returns></returns>
public virtual RTCRtpTransceiver AddTrack(string connectionId, MediaStreamTrack track)
{
return(m_handler.AddTrack(connectionId, track));
}
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(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($"dtls handshake result {t.Result}.");
if (t.Result)
{
//var remoteEP = peerConnection.IceSession.ConnectedRemoteEndPoint;
//peerConnection.SetDestination(SDPMediaTypesEnum.audio, remoteEP, remoteEP);
}
else
{
dtls.Shutdown();
peerConnection.Close("dtls handshake failed.");
}
});
var offerSdp = peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerSdp);
_peerConnections.TryAdd(id, peerConnection);
return(offerSdp);
}
/// <summary>
///
/// </summary>
/// <param name="connectionId"></param>
/// <param name="track"></param>
public virtual void RemoveTrack(string connectionId, MediaStreamTrack track)
{
m_handler.RemoveTrack(connectionId, track);
}
private static RTCPeerConnection CreatePeerConnection(WebSocketContext context)
{
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
},
X_RemoteSignallingAddress = context.UserEndPoint.Address,
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = SIPSORCERY_STUN_SERVER
}
}
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
// Add inactive audio and video tracks.
MediaStreamTrack audioTrack = new MediaStreamTrack("0", SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.Inactive);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
}, MediaStreamStatusEnum.Inactive);
peerConnection.addTrack(videoTrack);
peerConnection.onicecandidate += (candidate) =>
{
logger.LogDebug($"ICE candidate discovered: {candidate}.");
// Host candidates are included in the SDP we send.
if (candidate.type != RTCIceCandidateType.host)
{
context.WebSocket.Send($"candidate:{candidate}");
}
};
peerConnection.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
};
peerConnection.oniceconnectionstatechange += (state) =>
{
logger.LogDebug($"ICE connection state change to {state}.");
if (state == RTCIceConnectionState.connected)
{
var remoteEndPoint = peerConnection.IceSession.NominatedCandidate.DestinationEndPoint;
logger.LogInformation($"ICE connected to remote end point {remoteEndPoint}.");
}
};
peerConnection.IceSession.StartGathering();
return(peerConnection);
}
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);
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(windowsVideoEndPoint.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
testPatternSource.OnVideoSourceRawSample += windowsVideoEndPoint.ExternalVideoSourceRawSample;
windowsVideoEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (sdpFormat) =>
windowsVideoEndPoint.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
await windowsVideoEndPoint.StartVideo();
await testPatternSource.StartVideo();
}
else if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await testPatternSource.CloseVideo();
await windowsVideoEndPoint.CloseVideo();
await audioSource.CloseAudio();
}
};
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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($"ICE connection state change to {state}.");
return(pc);
}
private static async Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(WEBCAM_NAME);
bool initResult = await winVideoEP.InitialiseVideoSourceDevice();
if (!initResult)
{
throw new ApplicationException("Could not initialise video capture device.");
}
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(false, 640, 480, 30);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(false, 1920, 1080, 30);
//await winVideoEP.InitialiseVideoSourceDevice();
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(winVideoEP.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
winVideoEP.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (videoFormats) =>
winVideoEP.SetVideoSourceFormat(videoFormats.First());
pc.OnAudioFormatsNegotiated += (audioFormats) =>
audioSource.SetAudioSourceFormat(audioFormats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"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);
}
public RTCRtpTransceiver AddTransceiver(int indexPeer, MediaStreamTrack track)
{
return(peers[indexPeer].AddTransceiver(track));
}
private static RTCPeerConnection CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
var mediaFileSource = new SIPSorceryMedia.FFmpeg.FFmpegFileSource(MP4_PATH, new AudioEncoder());
mediaFileSource.Initialise();
mediaFileSource.RestrictCodecs(new List <VideoCodecsEnum> {
VideoCodecsEnum.VP8
});
mediaFileSource.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
mediaFileSource.OnEndOfFile += () => pc.Close("source eof");
MediaStreamTrack videoTrack = new MediaStreamTrack(mediaFileSource.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(mediaFileSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
mediaFileSource.OnVideoSourceEncodedSample += pc.SendVideo;
mediaFileSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (sdpFormat) => mediaFileSource.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.OnAudioFormatsNegotiated += (sdpFormat) => mediaFileSource.SetAudioSourceFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await mediaFileSource.CloseVideo();
}
else if (state == RTCPeerConnectionState.connected)
{
await mediaFileSource.StartVideo();
}
};
// Diagnostics.
//pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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);
}
