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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
private static RTCPeerConnection WebSocketOpened(WebSocketContext context)
{
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_CertificatePath = DTLS_CERTIFICATE_PATH,
X_KeyPath = DTLS_KEY_PATH,
X_Fingerprint = DTLS_CERTIFICATE_FINGERPRINT
}
}
};
var peerConnection = new RTCPeerConnection(pcConfiguration);
// Add local recvonly tracks. This ensures that the SDP answer includes only
// the codecs we support.
MediaStreamTrack audioTrack = new MediaStreamTrack("0", SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
return(peerConnection);
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var pc = new RTCPeerConnection(null);
MediaStreamTrack audioTrack = new MediaStreamTrack(new List <AudioFormat> {
new AudioFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
MediaStreamTrack videoTrack = new MediaStreamTrack(new List <VideoFormat> {
new VideoFormat(VideoCodecsEnum.VP8, 100)
});
pc.addTrack(audioTrack);
pc.addTrack(videoTrack);
pc.onconnectionstatechange += (state) =>
{
Log.LogDebug($"Peer connection state change to {state}.");
RequestSIPAgentKeyFrame(_rtpSession);
};
pc.OnRtpPacketReceived += ForwardAudioToSIP;
pc.OnVideoFrameReceived += ForwardVideoFrameToSIP; // ForwardVideoFrameToPeerConnection;
_peerConnection = pc;
return(Task.FromResult(pc));
}
private static Task <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, false, new AudioEncoder());
mediaFileSource.Initialise();
mediaFileSource.RestrictFormats(x => x.Codec == VideoCodecsEnum.VP8);
mediaFileSource.RestrictFormats(x => x.Codec == 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 += (videoFormats) => mediaFileSource.SetVideoSourceFormat(videoFormats.First());
pc.OnAudioFormatsNegotiated += (audioFormats) => mediaFileSource.SetAudioSourceFormat(audioFormats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await mediaFileSource.CloseVideo();
}
else if (state == RTCPeerConnectionState.connected)
{
await mediaFileSource.StartVideo();
}
};
// Diagnostics.
//pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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(Task.FromResult(pc));
}
private static async Task WebSocketMessageReceived(WebSocketContext context, RTCPeerConnection peerConnection, string message)
{
try
{
if (peerConnection.localDescription == null)
{
logger.LogDebug("Offer SDP: " + message);
// Add local media tracks depending on what was offered. Also add local tracks with the same media ID as
// the remote tracks so that the media announcement in the SDP answer are in the same order.
SDP remoteSdp = SDP.ParseSDPDescription(message);
var remoteAudioAnn = remoteSdp.Media.Where(x => x.Media == SDPMediaTypesEnum.audio).FirstOrDefault();
var remoteVideoAnn = remoteSdp.Media.Where(x => x.Media == SDPMediaTypesEnum.video).FirstOrDefault();
if (remoteAudioAnn != null)
{
MediaStreamTrack audioTrack = new MediaStreamTrack(remoteAudioAnn.MediaID, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
}
if (remoteVideoAnn != null)
{
MediaStreamTrack videoTrack = new MediaStreamTrack(remoteVideoAnn.MediaID, SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
}
// After local media tracks have been added the remote description can be set.
await peerConnection.setRemoteDescription(new RTCSessionDescriptionInit { sdp = message, type = RTCSdpType.offer });
var answer = await peerConnection.createAnswer(null);
await peerConnection.setLocalDescription(answer);
context.WebSocket.Send(answer.sdp);
}
else if (peerConnection.remoteDescription == null)
{
logger.LogDebug("Answer SDP: " + message);
await peerConnection.setRemoteDescription(new RTCSessionDescriptionInit { sdp = message, type = RTCSdpType.answer });
}
else
{
logger.LogDebug("ICE Candidate: " + message);
await peerConnection.addIceCandidate(new RTCIceCandidateInit { candidate = message });
}
}
catch (Exception excp)
{
logger.LogError("Exception WebSocketMessageReceived. " + excp.Message);
}
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var peerConnection = new RTCPeerConnection(null);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendOnly);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
}, MediaStreamStatusEnum.SendOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) => peerConnection.Close("remote timeout");
peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
peerConnection.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
OnVideoSampleReady -= peerConnection.SendVideo;
OnAudioSampleReady -= peerConnection.SendAudio;
peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
peerConnection.OnSendReport -= RtpSession_OnSendReport;
}
else if (state == RTCPeerConnectionState.connected)
{
if (!_isSampling)
{
_isSampling = true;
OnVideoSampleReady += peerConnection.SendVideo;
OnAudioSampleReady += peerConnection.SendAudio;
_ = Task.Run(StartMedia);
}
}
};
var offerInit = peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerInit);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
context.WebSocket.Send(offerInit.sdp);
return(peerConnection);
}
private static Task <RTCPeerConnection> CreatePeerConnection(X509Certificate2 cert)
{
//RTCConfiguration config = new RTCConfiguration
//{
// iceServers = new List<RTCIceServer> { new RTCIceServer { urls = STUN_URL } },
// certificates = new List<RTCCertificate> { new RTCCertificate { Certificate = cert } }
//};
//var pc = new RTCPeerConnection(config);
var pc = new RTCPeerConnection(null);
var testPatternSource = new VideoTestPatternSource(new SIPSorceryMedia.Encoders.VideoEncoder());
testPatternSource.SetFrameRate(60);
//testPatternSource.SetMaxFrameRate(true);
//var videoEndPoint = new SIPSorceryMedia.FFmpeg.FFmpegVideoEndPoint();
//videoEndPoint.RestrictFormats(format => format.Codec == VideoCodecsEnum.H264);
//var videoEndPoint = new SIPSorceryMedia.Windows.WindowsEncoderEndPoint();
//var videoEndPoint = new SIPSorceryMedia.Encoders.VideoEncoderEndPoint();
MediaStreamTrack track = new MediaStreamTrack(testPatternSource.GetVideoSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(track);
//testPatternSource.OnVideoSourceRawSample += videoEndPoint.ExternalVideoSourceRawSample;
testPatternSource.OnVideoSourceRawSample += MesasureTestPatternSourceFrameRate;
testPatternSource.OnVideoSourceEncodedSample += pc.SendVideo;
pc.OnVideoFormatsNegotiated += (formats) => testPatternSource.SetVideoSourceFormat(formats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
await testPatternSource.CloseVideo();
await testPatternSource.CloseVideo();
testPatternSource.Dispose();
}
else if (state == RTCPeerConnectionState.connected)
{
await testPatternSource.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(Task.FromResult(pc));
}
private static RTCPeerConnection CreatePeerConnection()
{
var pc = new RTCPeerConnection(new RTCConfiguration {
X_UseRtpFeedbackProfile = true
});
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPWellKnownMediaFormatsEnum.PCMU);
pc.addTrack(audioTrack);
pc.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
pc.OnTimeout += (mediaType) => logger.LogWarning($"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.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.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
pc.onsignalingstatechange += () =>
{
if (pc.signalingState == RTCSignalingState.have_remote_offer)
{
logger.LogTrace("Remote SDP:");
logger.LogTrace(pc.remoteDescription.sdp.ToString());
}
else if (pc.signalingState == RTCSignalingState.have_local_offer)
{
logger.LogTrace("Local SDP:");
logger.LogTrace(pc.localDescription.sdp.ToString());
}
};
return(pc);
}
public void JsonRoundtripUnitTest()
{
RTCPeerConnection pcSrc = new RTCPeerConnection(null);
var videoTrackSrc = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPMediaTypesEnum.video, 96, "VP8", 90000)
});
pcSrc.addTrack(videoTrackSrc);
var offer = pcSrc.createOffer(new RTCOfferOptions());
Assert.NotNull(offer.toJSON());
logger.LogDebug($"offer: {offer.toJSON()}");
var parseResult = RTCSessionDescriptionInit.TryParse(offer.toJSON(), out var init);
Assert.True(parseResult);
Assert.Equal(RTCSdpType.offer, init.type);
Assert.NotNull(init.sdp);
SDP sdp = SDP.ParseSDPDescription(init.sdp);
Assert.Equal(0, sdp.Version);
}
private static Task <RTCPeerConnection> CreatePeerConnection(IVideoSource videoSource, IVideoSink videoSink)
{
var pc = new RTCPeerConnection(null);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoSink.GetVideoSinkFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
videoSource.OnVideoSourceEncodedSample += pc.SendVideo;
pc.OnVideoFrameReceived += videoSink.GotVideoFrame;
pc.OnVideoFormatsNegotiated += (formats) =>
{
videoSink.SetVideoSinkFormat(formats.First());
videoSource.SetVideoSourceFormat(formats.First());
};
pc.OnTimeout += (mediaType) => logger.LogDebug($"Timeout on media {mediaType}.");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection connected changed to {state}.");
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.failed)
{
await videoSource.CloseVideo().ConfigureAwait(false);
}
};
return(Task.FromResult(pc));
}
public void GenerateLocalOfferWithAudioTrackUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTCPeerConnection pc = new RTCPeerConnection(null);
pc.IceSession.StartGathering();
var audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
pc.addTrack(audioTrack);
var offer = pc.createOffer(new RTCOfferOptions());
SDP offerSDP = SDP.ParseSDPDescription(offer.sdp);
Assert.NotNull(offer);
Assert.NotNull(offer.sdp);
Assert.Equal(RTCSdpType.offer, offer.type);
Assert.Single(offerSDP.Media);
Assert.Contains(offerSDP.Media, x => x.Media == SDPMediaTypesEnum.audio);
logger.LogDebug(offer.sdp);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
Log.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
_peerConnection = new RTCPeerConnection(null);
_peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
_peerConnection.OnSendReport += RtpSession_OnSendReport;
Log.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
_peerConnection.onconnectionstatechange += (state) =>
{
Log.LogDebug($"WebRTC peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.closed)
{
_peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
_peerConnection.OnSendReport -= RtpSession_OnSendReport;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
_peerConnection.addTrack(audioTrack);
var offerInit = _peerConnection.createOffer(null);
await _peerConnection.setLocalDescription(offerInit);
context.WebSocket.Send(offerInit.sdp);
return(_peerConnection);
}
public async void CheckPeerConnectionEstablishment()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
var aliceConnected = new TaskCompletionSource <bool>(TaskCreationOptions.RunContinuationsAsynchronously);
var bobConnected = new TaskCompletionSource <bool>(TaskCreationOptions.RunContinuationsAsynchronously);
var alice = new RTCPeerConnection();
alice.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Alice connected.");
aliceConnected.SetResult(true);
}
};
alice.addTrack(new MediaStreamTrack(SDPWellKnownMediaFormatsEnum.PCMU));
var aliceOffer = alice.createOffer();
await alice.setLocalDescription(aliceOffer);
logger.LogDebug($"alice offer: {aliceOffer.sdp}");
var bob = new RTCPeerConnection();
bob.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Bob connected.");
bobConnected.SetResult(true);
}
};
bob.addTrack(new MediaStreamTrack(SDPWellKnownMediaFormatsEnum.PCMU));
var setOfferResult = bob.setRemoteDescription(aliceOffer);
Assert.Equal(SetDescriptionResultEnum.OK, setOfferResult);
var bobAnswer = bob.createAnswer();
await bob.setLocalDescription(bobAnswer);
var setAnswerResult = alice.setRemoteDescription(bobAnswer);
Assert.Equal(SetDescriptionResultEnum.OK, setAnswerResult);
logger.LogDebug($"answer: {bobAnswer.sdp}");
await Task.WhenAny(Task.WhenAll(aliceConnected.Task, bobConnected.Task), Task.Delay(2000));
Assert.True(aliceConnected.Task.IsCompleted);
Assert.True(aliceConnected.Task.Result);
Assert.True(bobConnected.Task.IsCompleted);
Assert.True(bobConnected.Task.Result);
bob.close();
alice.close();
}
private static async Task WebSocketMessageReceived(WebSocketContext context, RTCPeerConnection pc, string message)
{
try
{
if (pc.localDescription == null)
{
//logger.LogDebug("Offer SDP: " + message);
logger.LogDebug("Offer SDP received.");
// Add local media tracks depending on what was offered. Also add local tracks with the same media ID as
// the remote tracks so that the media announcement in the SDP answer are in the same order.
SDP remoteSdp = SDP.ParseSDPDescription(message);
foreach (var ann in remoteSdp.Media)
{
var capbilities = FilterCodecs(ann.Media, ann.MediaFormats);
MediaStreamTrack track = new MediaStreamTrack(ann.Media, false, capbilities, MediaStreamStatusEnum.RecvOnly);
pc.addTrack(track);
}
pc.setRemoteDescription(new RTCSessionDescriptionInit {
sdp = message, type = RTCSdpType.offer
});
var answer = pc.createAnswer(null);
await pc.setLocalDescription(answer);
Console.WriteLine(answer.sdp);
context.WebSocket.Send(answer.sdp);
}
else if (pc.remoteDescription == null)
{
logger.LogDebug("Answer SDP: " + message);
pc.setRemoteDescription(new RTCSessionDescriptionInit {
sdp = message, type = RTCSdpType.answer
});
}
else
{
logger.LogDebug("ICE Candidate: " + message);
if (string.IsNullOrWhiteSpace(message) || message.Trim().ToLower() == SDP.END_ICE_CANDIDATES_ATTRIBUTE)
{
logger.LogDebug("End of candidates message received.");
}
else
{
var candInit = Newtonsoft.Json.JsonConvert.DeserializeObject <RTCIceCandidateInit>(message);
pc.addIceCandidate(candInit);
}
}
}
catch (Exception excp)
{
logger.LogError("Exception WebSocketMessageReceived. " + excp.Message);
}
}
static void Main()
{
Console.WriteLine("SIPSorcery sip.js Demo");
Log = AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var sipChannel = new SIPWebSocketChannel(IPAddress.Loopback, 8081);
sipTransport.AddSIPChannel(sipChannel);
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogDebug($"Auto-answering incoming call from {req.Header.From}.");
var uas = userAgent.AcceptCall(req);
var peerConnection = new RTCPeerConnection(null);
peerConnection.onconnectionstatechange += (state) =>
{
Log.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
peerConnection.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.connected)
{
peerConnection.OnRtpPacketReceived += OnRtpPacketReceived;
}
else if (state == RTCPeerConnectionState.closed)
{
peerConnection.OnRtpPacketReceived -= OnRtpPacketReceived;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(new List <AudioFormat> {
new AudioFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendRecv);
peerConnection.addTrack(audioTrack);
//MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, MediaStreamStatusEnum.Inactive);
//peerConnection.addTrack(videoTrack);
var answerResult = await userAgent.Answer(uas, peerConnection);
};
Console.Write("press any key to exit...");
Console.Read();
sipTransport.Shutdown();
}
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 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($"ICE connection state change to {state}.");
return(pc);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var pc = new RTCPeerConnection(null);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
}, MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
pc.OnReceiveReport += RtpSession_OnReceiveReport;
pc.OnSendReport += RtpSession_OnSendReport;
pc.OnTimeout += (mediaType) => pc.Close("remote timeout");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
pc.Close("remote disconnection");
}
if (state == RTCPeerConnectionState.closed)
{
OnTestPatternSampleReady -= pc.SendMedia;
pc.OnReceiveReport -= RtpSession_OnReceiveReport;
pc.OnSendReport -= RtpSession_OnSendReport;
_sendTestPatternTimer?.Dispose();
}
else if (state == RTCPeerConnectionState.connected)
{
OnTestPatternSampleReady += pc.SendMedia;
if (_sendTestPatternTimer == null)
{
_sendTestPatternTimer = new Timer(SendTestPattern, null, 0, TEST_PATTERN_SPACING_MILLISECONDS);
}
}
};
var offerSdp = pc.createOffer(null);
await pc.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
logger.LogDebug(offerSdp.sdp);
context.WebSocket.Send(offerSdp.sdp);
return(pc);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var pc = new RTCPeerConnection(null);
AudioExtrasSource audioSource = new AudioExtrasSource(new AudioEncoder());
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.OnReceiveReport += RtpSession_OnReceiveReport;
pc.OnSendReport += RtpSession_OnSendReport;
pc.OnTimeout += (mediaType) => pc.Close("remote timeout");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
audioSource.SetSource(new AudioSourceOptions {
AudioSource = AudioSourcesEnum.SineWave
});
}
else if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
pc.Close("remote disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
audioSource?.CloseAudio();
pc.OnReceiveReport -= RtpSession_OnReceiveReport;
pc.OnSendReport -= RtpSession_OnSendReport;
}
};
var offerSdp = pc.createOffer(null);
await pc.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
logger.LogDebug(offerSdp.sdp);
context.WebSocket.Send(offerSdp.sdp);
return(pc);
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
AudioExtrasSource audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.SineWave
});
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
pc.OnAudioFormatsNegotiated += (sdpFormat) =>
audioSource.SetAudioSourceFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"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));
}
private RTCPeerConnection CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration();
RTCPeerConnection pc = new RTCPeerConnection(config);
MediaStreamTrack videoTrack = new MediaStreamTrack(new List <VideoCodecsEnum>()
{
VideoCodecsEnum.H264
});
pc.addTrack(videoTrack);
pc.OnVideoFormatsNegotiated += Pc_OnVideoFormatsNegotiated;
ConnectedPeer = pc;
return(pc);
}
private Task <RTCPeerConnection> CreatePeerConnection()
{
var pc = new RTCPeerConnection();
// Set up sources and hook up send events to peer connection.
//AudioExtrasSource audioSrc = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.None });
//audioSrc.OnAudioSourceEncodedSample += pc.SendAudio;
//var testPatternSource = new VideoTestPatternSource();
//testPatternSource.SetMaxFrameRate(true);
//testPatternSource.OnVideoSourceRawSample += VideoEncoderEndPoint.ExternalVideoSourceRawSample;
//VideoEncoderEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
// Add tracks.
//var audioTrack = new SIPSorcery.Net.MediaStreamTrack(audioSrc.GetAudioSourceFormats(), SIPSorcery.Net.MediaStreamStatusEnum.SendOnly);
//pc.addTrack(audioTrack);
var videoTrack = new MediaStreamTrack(VideoEncoderEndPoint.GetVideoSourceFormats(), MediaStreamStatusEnum.RecvOnly);
pc.addTrack(videoTrack);
// Handlers to set the codecs to use on the sources once the SDP negotiation is complete.
pc.OnVideoFormatsNegotiated += (sdpFormat) => VideoEncoderEndPoint.SetVideoSourceFormat(sdpFormat.First());
//pc.OnAudioFormatsNegotiated += (sdpFormat) => audioSrc.SetAudioSourceFormat(sdpFormat.First());
pc.OnVideoFrameReceived += VideoEncoderEndPoint.GotVideoFrame;
pc.OnTimeout += (mediaType) => logger.LogDebug($"Timeout on media {mediaType}.");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection connected changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
//await audioSrc.StartAudio();
//await testPatternSource.StartVideo();
}
else if (state == RTCPeerConnectionState.closed || state == SIPSorcery.Net.RTCPeerConnectionState.failed)
{
//await audioSrc.CloseAudio();
//await testPatternSource.CloseVideo();
}
};
//pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
//{
// bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == SIPSorcery.Net.STUNAttributeTypesEnum.UseCandidate);
// logger.LogDebug($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
//};
return(Task.FromResult(pc));
}
public void SendVideoRtcpFeedbackReportUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
Certificate = DtlsUtils.CreateSelfSignedCert()
}
},
X_UseRtpFeedbackProfile = true
};
RTCPeerConnection pcSrc = new RTCPeerConnection(pcConfiguration);
var videoTrackSrc = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
pcSrc.addTrack(videoTrackSrc);
var offer = pcSrc.createOffer(new RTCOfferOptions());
logger.LogDebug($"offer: {offer.sdp}");
RTCPeerConnection pcDst = new RTCPeerConnection(pcConfiguration);
var videoTrackDst = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
pcDst.addTrack(videoTrackDst);
var setOfferResult = pcDst.setRemoteDescription(offer);
Assert.Equal(SetDescriptionResultEnum.OK, setOfferResult);
var answer = pcDst.createAnswer(null);
var setAnswerResult = pcSrc.setRemoteDescription(answer);
Assert.Equal(SetDescriptionResultEnum.OK, setAnswerResult);
logger.LogDebug($"answer: {answer.sdp}");
RTCPFeedback pliReport = new RTCPFeedback(pcDst.VideoLocalTrack.Ssrc, pcDst.VideoRemoteTrack.Ssrc, PSFBFeedbackTypesEnum.PLI);
pcDst.SendRtcpFeedback(SDPMediaTypesEnum.video, pliReport);
}
public void SendVideoRtcpFeedbackReportUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = new List <RTCCertificate>
{
new RTCCertificate
{
X_Fingerprint = "sha-256 C6:ED:8C:9D:06:50:77:23:0A:4A:D8:42:68:29:D0:70:2F:BB:C7:72:EC:98:5C:62:07:1B:0C:5D:CB:CE:BE:CD"
}
},
X_UseRtpFeedbackProfile = true
};
RTCPeerConnection pcSrc = new RTCPeerConnection(pcConfiguration);
var videoTrackSrc = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
pcSrc.addTrack(videoTrackSrc);
var offer = pcSrc.createOffer(new RTCOfferOptions());
logger.LogDebug($"offer: {offer.sdp}");
RTCPeerConnection pcDst = new RTCPeerConnection(pcConfiguration);
var videoTrackDst = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
});
pcDst.addTrack(videoTrackDst);
var setOfferResult = pcDst.setRemoteDescription(offer);
Assert.Equal(SetDescriptionResultEnum.OK, setOfferResult);
var answer = pcDst.createAnswer(null);
var setAnswerResult = pcSrc.setRemoteDescription(answer);
Assert.Equal(SetDescriptionResultEnum.OK, setAnswerResult);
logger.LogDebug($"answer: {answer.sdp}");
RTCPFeedback pliReport = new RTCPFeedback(pcDst.VideoLocalTrack.Ssrc, pcDst.VideoRemoteTrack.Ssrc, PSFBFeedbackTypesEnum.PLI);
pcDst.SendRtcpFeedback(SDPMediaTypesEnum.video, pliReport);
}
private static RTCPeerConnection WebSocketOpened(WebSocketContext context)
{
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(SDPMediaTypesEnum.video, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.VP8)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.oniceconnectionstatechange += (state) => Console.WriteLine($"ICE connection state changed to {state}.");
peerConnection.onconnectionstatechange += (state) =>
{
Console.WriteLine($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.closed)
{
peerConnection.OnRtpPacketReceived -= RtpSession_OnRtpPacketReceived;
}
else if (state == RTCPeerConnectionState.connected)
{
peerConnection.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
}
};
return(peerConnection);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
Log.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
_peerConnection = new RTCPeerConnection(null);
//AddressFamily.InterNetwork,
//DTLS_CERTIFICATE_FINGERPRINT,
//null,
//null);
_peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
_peerConnection.OnSendReport += RtpSession_OnSendReport;
Log.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
_peerConnection.onconnectionstatechange += (state) =>
{
if (state == RTCPeerConnectionState.closed)
{
Log.LogDebug($"RTC peer connection closed.");
_peerConnection.OnReceiveReport -= RtpSession_OnReceiveReport;
_peerConnection.OnSendReport -= RtpSession_OnSendReport;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
_peerConnection.addTrack(audioTrack);
var offerInit = await _peerConnection.createOffer(null);
await _peerConnection.setLocalDescription(offerInit);
context.WebSocket.Send(offerInit.sdp);
if (DoDtlsHandshake(_peerConnection))
{
Log.LogInformation("DTLS handshake completed successfully.");
}
else
{
_peerConnection.Close("dtls handshake failed.");
}
return(_peerConnection);
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var pc = new RTCPeerConnection(null);
MediaStreamTrack track = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
pc.addTrack(track);
pc.onconnectionstatechange += (state) => Log.LogDebug($"Peer connection state change to {state}.");
pc.OnRtpPacketReceived += ForwardMediaToSIP;
_peerConnection = pc;
return(Task.FromResult(pc));
}
private static RTCPeerConnection Createpc(WebSocketContext context, SDPAudioVideoMediaFormat videoFormat)
{
var pc = new RTCPeerConnection(null);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.video, false, new List <SDPAudioVideoMediaFormat> {
videoFormat
}, MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
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.OnRtpClosed += (reason) => logger.LogDebug($"Peer connection closed, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
pc.onicecandidate += (candidate) =>
{
if (pc.signalingState == RTCSignalingState.have_local_offer ||
pc.signalingState == RTCSignalingState.have_remote_offer)
{
context.WebSocket.Send($"candidate:{candidate}");
}
};
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
logger.LogDebug("Creating RTP session to receive ffmpeg stream.");
_ffmpegListener.OnRtpPacketReceived += (ep, media, rtpPkt) =>
{
if (media == SDPMediaTypesEnum.video && pc.VideoDestinationEndPoint != null)
{
//logger.LogDebug($"Forwarding {media} RTP packet to webrtc peer timestamp {rtpPkt.Header.Timestamp}.");
pc.SendRtpRaw(media, rtpPkt.Payload, rtpPkt.Header.Timestamp, rtpPkt.Header.MarkerBit, rtpPkt.Header.PayloadType);
}
};
}
};
return(pc);
}
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.");
}
var peerConnection = new RTCPeerConnection(null);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) => _logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}, SeqNum {rtpPkt.Header.SequenceNumber}.");
//peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
//peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) => _logger.LogWarning($"Timeout for {mediaType}.");
peerConnection.onconnectionstatechange += (state) =>
{
_logger.LogDebug($"Peer connection {id} state changed to {state}.");
if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
_peerConnections.TryRemove(id, out _);
}
else if (state == RTCPeerConnectionState.connected)
{
_logger.LogDebug("Peer connection connected.");
}
};
var offerSdp = peerConnection.createOffer(null);
await peerConnection.setLocalDescription(offerSdp);
_peerConnections.TryAdd(id, peerConnection);
return(offerSdp);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"Web socket client connection from {context.UserEndPoint}.");
var pc = new RTCPeerConnection(null);
MediaStreamTrack videoTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
pc.OnReceiveReport += RtpSession_OnReceiveReport;
pc.OnSendReport += RtpSession_OnSendReport;
pc.OnTimeout += (mediaType) => pc.Close("remote timeout");
pc.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state change to {state}.");
pc.onconnectionstatechange += (state) =>
{
logger.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
pc.Close("remote disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
pc.OnReceiveReport -= RtpSession_OnReceiveReport;
pc.OnSendReport -= RtpSession_OnSendReport;
}
};
var offerSdp = pc.createOffer(null);
await pc.setLocalDescription(offerSdp);
logger.LogDebug($"Sending SDP offer to client {context.UserEndPoint}.");
logger.LogDebug(offerSdp.sdp);
context.WebSocket.Send(offerSdp.sdp);
return(pc);
}