private void AudioFormatsNegotiated(List <SDPMediaFormat> audoFormats)
{
var audioCodec = SDPMediaFormatInfo.GetAudioCodecForSdpFormat(audoFormats.First().FormatCodec);
logger.LogDebug($"Setting audio source format to {audioCodec}.");
AudioExtrasSource.SetAudioSourceFormat(audioCodec);
}
private static async Task <RTCPeerConnection> SendSDPOffer(WebSocketContext context)
{
logger.LogDebug($"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 Task <RTCPeerConnection> CreatePeerConnection(string url)
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
//mediaFileSource.OnEndOfFile += () => pc.Close("source eof");
MediaStreamTrack videoTrack = new MediaStreamTrack(new List <VideoCodecsEnum> {
VIDEO_CODEC
}, MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(new List <AudioCodecsEnum> {
AUDIO_CODEC
}, MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
IVideoSource videoSource = null;
IAudioSource audioSource = null;
if (url == MAX_URL)
{
videoSource = _maxSource;
audioSource = _maxSource;
}
else
{
videoSource = _testPatternEncoder;
audioSource = _musicSource;
}
pc.OnVideoFormatsNegotiated += (sdpFormat) => videoSource.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.OnAudioFormatsNegotiated += (sdpFormat) => audioSource.SetAudioSourceFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
videoSource.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.onconnectionstatechange += async(state) =>
{
_logger.LogInformation($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
pc.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.closed)
{
videoSource.OnVideoSourceEncodedSample -= pc.SendVideo;
audioSource.OnAudioSourceEncodedSample -= pc.SendAudio;
await CheckForSourceSubscribers();
}
else if (state == RTCPeerConnectionState.connected)
{
await StartSource(url);
}
};
// Diagnostics.
//pc.OnReceiveReport += (re, media, rr) => logger.LogDebug($"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.LogInformation($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
private static Task <RTCPeerConnection> CreatePeerConnection()
{
var peerConnection = new RTCPeerConnection(null);
var videoEP = new SIPSorceryMedia.Encoders.VideoEncoderEndPoint();
//var videoEP = new SIPSorceryMedia.Windows.WindowsEncoderEndPoint();
//var videoEP = new FFmpegVideoEndPoint();
videoEP.RestrictCodecs(new List <VideoCodecsEnum> {
VideoCodecsEnum.VP8
});
videoEP.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
_form.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = bmp)
{
Bitmap bmpImage = new Bitmap((int)width, (int)height, (int)(bmp.Length / height), PixelFormat.Format24bppRgb, (IntPtr)s);
_picBox.Image = bmpImage;
}
}
}));
};
// Sink (speaker) only audio end point.
WindowsAudioEndPoint windowsAudioEP = new WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(windowsAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
MediaStreamTrack videoTrack = new MediaStreamTrack(videoEP.GetVideoSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(videoTrack);
peerConnection.OnVideoFrameReceived += videoEP.GotVideoFrame;
peerConnection.OnVideoFormatsNegotiated += (sdpFormat) => videoEP.SetVideoSinkFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
peerConnection.OnAudioFormatsNegotiated += (sdpFormat) =>
windowsAudioEP.SetAudioSinkFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"Timeout on media {mediaType}.");
peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"ICE connection state changed to {state}.");
peerConnection.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection connected changed to {state}.");
if (state == RTCPeerConnectionState.connected)
{
await windowsAudioEP.StartAudio();
}
else if (state == RTCPeerConnectionState.closed || state == RTCPeerConnectionState.failed)
{
await windowsAudioEP.CloseAudio();
}
};
peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == STUNAttributeTypesEnum.UseCandidate);
Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
peerConnection.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
//logger.LogDebug($"RTP {media} pkt received, SSRC {rtpPkt.Header.SyncSource}.");
if (media == SDPMediaTypesEnum.audio)
{
windowsAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
}
};
return(Task.FromResult(peerConnection));
}
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.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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
private static Task <RTCPeerConnection> Createpc(WebSocketContext context, RTCIceServer stunServer, bool relayOnly)
{
if (_peerConnection != null)
{
_peerConnection.Close("normal");
}
List <RTCCertificate> presetCertificates = null;
if (File.Exists(LOCALHOST_CERTIFICATE_PATH))
{
var localhostCert = new X509Certificate2(LOCALHOST_CERTIFICATE_PATH, (string)null, X509KeyStorageFlags.Exportable);
presetCertificates = new List <RTCCertificate> {
new RTCCertificate {
Certificate = localhostCert
}
};
}
RTCConfiguration pcConfiguration = new RTCConfiguration
{
certificates = presetCertificates,
//X_RemoteSignallingAddress = (context != null) ? context.UserEndPoint.Address : null,
iceServers = stunServer != null ? new List <RTCIceServer> {
stunServer
} : null,
//iceTransportPolicy = RTCIceTransportPolicy.all,
iceTransportPolicy = relayOnly ? RTCIceTransportPolicy.relay : RTCIceTransportPolicy.all,
//X_BindAddress = IPAddress.Any, // NOTE: Not reqd. Using this to filter out IPv6 addresses so can test with Pion.
};
_peerConnection = new RTCPeerConnection(pcConfiguration);
//_peerConnection.GetRtpChannel().MdnsResolve = (hostname) => Task.FromResult(NetServices.InternetDefaultAddress);
_peerConnection.GetRtpChannel().MdnsResolve = MdnsResolve;
//_peerConnection.GetRtpChannel().OnStunMessageReceived += (msg, ep, isrelay) => logger.LogDebug($"STUN message received from {ep}, message type {msg.Header.MessageType}.");
var dc = _peerConnection.createDataChannel(DATA_CHANNEL_LABEL, null);
dc.onmessage += (msg) => logger.LogDebug($"data channel receive ({dc.label}-{dc.id}): {msg}");
// Add a send-only audio track (this doesn't require any native libraries for encoding so is good for x-platform testing).
AudioExtrasSource audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.SineWave
});
audioSource.OnAudioSourceEncodedSample += _peerConnection.SendAudio;
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
_peerConnection.addTrack(audioTrack);
_peerConnection.OnAudioFormatsNegotiated += (sdpFormat) =>
audioSource.SetAudioSourceFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
_peerConnection.onicecandidateerror += (candidate, error) => logger.LogWarning($"Error adding remote ICE candidate. {error} {candidate}");
_peerConnection.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"Peer connection state changed to {state}.");
if (state == RTCPeerConnectionState.disconnected || state == RTCPeerConnectionState.failed)
{
_peerConnection.Close("remote disconnection");
}
if (state == RTCPeerConnectionState.connected)
{
await audioSource.StartAudio();
}
else if (state == RTCPeerConnectionState.closed)
{
await audioSource.CloseAudio();
}
};
_peerConnection.OnReceiveReport += (ep, type, rtcp) => logger.LogDebug($"RTCP {type} report received.");
_peerConnection.OnRtcpBye += (reason) => logger.LogDebug($"RTCP BYE receive, reason: {(string.IsNullOrWhiteSpace(reason) ? "<none>" : reason)}.");
_peerConnection.onicecandidate += (candidate) =>
{
if (_peerConnection.signalingState == RTCSignalingState.have_local_offer ||
_peerConnection.signalingState == RTCSignalingState.have_remote_offer)
{
if (context != null &&
(_iceTypes.Count == 0 || _iceTypes.Any(x => x == candidate.type)))
{
var candidateInit = new RTCIceCandidateInit
{
sdpMid = candidate.sdpMid,
sdpMLineIndex = candidate.sdpMLineIndex,
usernameFragment = candidate.usernameFragment,
candidate = "candidate:" + candidate.ToString()
};
context.WebSocket.Send(JsonConvert.SerializeObject(candidateInit));
}
}
};
// Peer ICE connection state changes are for ICE events such as the STUN checks completing.
_peerConnection.oniceconnectionstatechange += (state) => logger.LogDebug($"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($"Web socket client connection from {context.UserEndPoint}.");
var peerConnection = new RTCPeerConnection(null);
// Sink (speaker) only audio end point.
WindowsAudioEndPoint windowsAudioEP = new WindowsAudioEndPoint(new AudioEncoder(), -1, -1, true, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(windowsAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.RecvOnly);
peerConnection.addTrack(audioTrack);
peerConnection.OnAudioFormatsNegotiated += (sdpFormat) =>
windowsAudioEP.SetAudioSinkFormat(SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
peerConnection.OnReceiveReport += RtpSession_OnReceiveReport;
peerConnection.OnSendReport += RtpSession_OnSendReport;
peerConnection.OnTimeout += (mediaType) => logger.LogDebug($"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);
}
