private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var peerConnection = new RTCPeerConnection(null);
// Sink (speaker) only audio end point.
WindowsAudioEndPoint windowsAudioEP = new WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(windowsAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
peerConnection.OnAudioFormatsNegotiated += (audioFormats) =>
windowsAudioEP.SetAudioSinkFormat(audioFormats.First());
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"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)
{
peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
peerConnection.OnSendReport -= RtpSession_OnSendReport;
await windowsAudioEP.CloseAudio();
}
};
peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
//logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
if (media == SDPMediaTypesEnum.audio)
{
windowsAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
}
};
var offerSdp = peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
logger.LogDebug(offerSdp.sdp);
context.WebSocket.Send(offerSdp.sdp);
return(peerConnection);
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
//AudioExtrasSource audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.SineWave });
//audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
WindowsAudioEndPoint audioSource = new WindowsAudioEndPoint(new AudioEncoder());
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
pc.OnAudioFormatsNegotiated += (audioFormats) => audioSource.SetAudioSourceFormat(audioFormats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
}
else if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await audioSource.CloseAudio();
}
};
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
pc.OnSendReport += (media, sr) => logger.LogDebug($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) => logger.LogDebug($"STUN {msg.Header.MessageType} received from {ep}.");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var peerConnection = new RTCPeerConnection(null);
var videoEP = new SIPSorceryMedia.Encoders.VideoEncoderEndPoint();
//var videoEP = new SIPSorceryMedia.Windows.WindowsEncoderEndPoint();
//var videoEP = new FFmpegVideoEndPoint();
videoEP.RestrictFormats(format => format.Codec == VideoCodecsEnum.VP8);
videoEP.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
_form.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = bmp)
{
Bitmap bmpImage = new Bitmap((int)width, (int)height, (int)(bmp.Length / height), PixelFormat.Format24bppRgb, (IntPtr)s);
_picBox.Image = bmpImage;
}
}
}));
};
// Sink (speaker) only audio end point.
WindowsAudioEndPoint windowsAudioEP = new WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(windowsAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnVideoFrameReceived += videoEP.GotVideoFrame;
peerConnection.OnVideoFormatsNegotiated += (formats) =>
videoEP.SetVideoSinkFormat(formats.First());
peerConnection.OnAudioFormatsNegotiated += (formats) =>
windowsAudioEP.SetAudioSinkFormat(formats.First());
peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"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));
}