private static async void MessageReceived(WebSocketContext context, string msg)
{
//Console.WriteLine($"websocket recv: {msg}");
var offerSDP = SDP.ParseSDPDescription(msg);
Console.WriteLine($"offer sdp: {offerSDP}");
var webRtcSession = new WebRtcSession(
AddressFamily.InterNetwork,
DTLS_CERTIFICATE_FINGERPRINT,
null,
null);
webRtcSession.setRemoteDescription(new RTCSessionDescription {
sdp = offerSDP, type = RTCSdpType.offer
});
webRtcSession.OnReceiveReport += RtpSession_OnReceiveReport;
webRtcSession.OnSendReport += RtpSession_OnSendReport;
webRtcSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
webRtcSession.OnClose += (reason) =>
{
Console.WriteLine($"webrtc session closed: {reason}");
_webRtcSessions.Remove(webRtcSession);
};
// Add local recvonly tracks. This ensures that the SDP answer includes only
// the codecs we support.
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
audioTrack.Transceiver.SetStreamStatus(MediaStreamStatusEnum.RecvOnly);
webRtcSession.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
videoTrack.Transceiver.SetStreamStatus(MediaStreamStatusEnum.RecvOnly);
webRtcSession.addTrack(videoTrack);
var answerSdp = await webRtcSession.createAnswer();
webRtcSession.setLocalDescription(new RTCSessionDescription {
sdp = answerSdp, type = RTCSdpType.answer
});
Console.WriteLine($"answer sdp: {answerSdp}");
context.WebSocket.Send(answerSdp.ToString());
if (DoDtlsHandshake(webRtcSession))
{
_webRtcSessions.Add(webRtcSession);
}
else
{
webRtcSession.Close("dtls handshake failed.");
}
}
private static async Task <WebRtcSession> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var webRtcSession = new WebRtcSession(
AddressFamily.InterNetwork,
DTLS_CERTIFICATE_FINGERPRINT,
null,
null);
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
webRtcSession.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
webRtcSession.addTrack(videoTrack);
//webRtcSession.RtpSession.OnReceiveReport += RtpSession_OnReceiveReport;
webRtcSession.OnSendReport += RtpSession_OnSendReport;
OnMediaSampleReady += webRtcSession.SendMedia;
webRtcSession.OnClose += (reason) =>
{
logger.LogDebug($"WebRtcSession was closed with reason {reason}");
OnMediaSampleReady -= webRtcSession.SendMedia;
webRtcSession.OnReceiveReport -= RtpSession_OnReceiveReport;
webRtcSession.OnSendReport -= RtpSession_OnSendReport;
};
var offerSdp = await webRtcSession.createOffer(null);
webRtcSession.setLocalDescription(new RTCSessionDescription {
sdp = offerSdp, type = RTCSdpType.offer
});
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
context.WebSocket.Send(offerSdp.ToString());
if (DoDtlsHandshake(webRtcSession))
{
if (!_isSampling)
{
_ = Task.Run(StartMedia);
}
}
else
{
webRtcSession.Close("dtls handshake failed.");
}
return(webRtcSession);
}
private static WebRtcSession SendSDPOffer(WebSocketContext context)
{
Log.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
_webRtcSession = new WebRtcSession(
AddressFamily.InterNetwork,
DTLS_CERTIFICATE_FINGERPRINT,
null,
null);
_webRtcSession.OnReceiveReport += RtpSession_OnReceiveReport;
_webRtcSession.OnSendReport += RtpSession_OnSendReport;
Log.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
_webRtcSession.OnClose += (reason) =>
{
Log.LogDebug($"WebRtcSession was closed with reason {reason}");
_webRtcSession.OnReceiveReport -= RtpSession_OnReceiveReport;
_webRtcSession.OnSendReport -= RtpSession_OnSendReport;
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
_webRtcSession.addTrack(audioTrack);
var offer = _webRtcSession.createOffer(null).Result;
_webRtcSession.setLocalDescription(new RTCSessionDescription {
type = RTCSdpType.offer, sdp = offer
});
context.WebSocket.Send(offer.ToString());
if (DoDtlsHandshake(_webRtcSession))
{
Log.LogInformation("DTLS handshake completed successfully.");
}
else
{
_webRtcSession.Close("dtls handshake failed.");
}
return(_webRtcSession);
}
private static async Task <WebRtcSession> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var webRtcSession = new WebRtcSession(
AddressFamily.InterNetwork,
DTLS_CERTIFICATE_FINGERPRINT,
null,
null);
webRtcSession.addTrack(SDPMediaTypesEnum.video, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
webRtcSession.RtpSession.OnReceiveReport += RtpSession_OnReceiveReport;
webRtcSession.RtpSession.OnSendReport += RtpSession_OnSendReport;
webRtcSession.OnClose += (reason) =>
{
logger.LogDebug($"WebRtcSession was closed with reason {reason}");
OnTestPatternSampleReady -= webRtcSession.SendMedia;
webRtcSession.RtpSession.OnReceiveReport -= RtpSession_OnReceiveReport;
webRtcSession.RtpSession.OnSendReport -= RtpSession_OnSendReport;
};
var offerSdp = await webRtcSession.createOffer();
webRtcSession.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
logger.LogDebug(webRtcSession.SDP.ToString());
context.WebSocket.Send(webRtcSession.SDP.ToString());
if (DoDtlsHandshake(webRtcSession))
{
OnTestPatternSampleReady += webRtcSession.SendMedia;
}
else
{
webRtcSession.Close("dtls handshake failed.");
}
return(webRtcSession);
}
private static WebRtcSession SendSDPOffer(WebSocketContext context)
{
Log.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var webRtcSession = new WebRtcSession(
AddressFamily.InterNetwork,
DTLS_CERTIFICATE_FINGERPRINT,
null,
null);
//webRtcSession.AudioStreamStatus = MediaStreamStatusEnum.SendOnly;
//webRtcSession.VideoStreamStatus = MediaStreamStatusEnum.SendOnly;
webRtcSession.RtpSession.OnReceiveReport += RtpSession_OnReceiveReport;
webRtcSession.RtpSession.OnSendReport += RtpSession_OnSendReport;
Log.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
webRtcSession.OnClose += (reason) =>
{
Log.LogDebug($"WebRtcSession was closed with reason {reason}");
OnMediaSampleReady -= webRtcSession.SendMedia;
webRtcSession.RtpSession.OnReceiveReport -= RtpSession_OnReceiveReport;
webRtcSession.RtpSession.OnSendReport -= RtpSession_OnSendReport;
};
webRtcSession.addTrack(SDPMediaTypesEnum.audio, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
if (DoDtlsHandshake(webRtcSession))
{
context.WebSocket.Send(webRtcSession.SDP.ToString());
}
else
{
webRtcSession.Close("dtls handshake failed.");
}
return(webRtcSession);
}
private static void SDPAnswerReceived(WebRtcSession webRtcSession, string sdpAnswer)
{
try
{
logger.LogDebug("Answer SDP: " + sdpAnswer);
var answerSDP = SDP.ParseSDPDescription(sdpAnswer);
webRtcSession.SdpSessionID = answerSDP.SessionId;
webRtcSession.RemoteIceUser = answerSDP.IceUfrag;
webRtcSession.RemoteIcePassword = answerSDP.IcePwd;
// All browsers seem to have gone to trickling ICE candidates now but just
// in case one or more are given we can start the STUN dance immediately.
if (answerSDP.IceCandidates != null)
{
foreach (var iceCandidate in answerSDP.IceCandidates)
{
webRtcSession.AppendRemoteIceCandidate(iceCandidate);
}
}
OnTestPatternSampleReady += (timestamp, sample) =>
{
try
{
webRtcSession.SendMedia(SDPMediaTypesEnum.video, timestamp, sample);
}
catch (Exception sendExcp)
{
logger.LogWarning("Exception OnTestPatternSampleReady. " + sendExcp.Message);
webRtcSession.Close();
}
};
}
catch (Exception excp)
{
logger.LogError("Exception SDPAnswerReceived. " + excp.Message);
}
}