private Task <SIPSorcery.Net.RTCPeerConnection> CreatePeerConnection()
{
var pc = new SIPSorcery.Net.RTCPeerConnection(null);
// 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 SIPSorcery.Net.MediaStreamTrack(VideoEncoderEndPoint.GetVideoSourceFormats(), SIPSorcery.Net.MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
// Handlers to set the codecs to use on the sources once the SDP negotiation is complete.
pc.OnVideoFormatsNegotiated += (sdpFormat) => VideoEncoderEndPoint.SetVideoSourceFormat(SIPSorcery.Net.SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.OnAudioFormatsNegotiated += (sdpFormat) =>
audioSrc.SetAudioSourceFormat(SIPSorcery.Net.SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
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($"Peer connection connected changed to {state}.");
if (state == SIPSorcery.Net.RTCPeerConnectionState.connected)
{
await audioSrc.StartAudio();
//await testPatternSource.StartVideo();
}
else if (state == SIPSorcery.Net.RTCPeerConnectionState.closed || state == SIPSorcery.Net.RTCPeerConnectionState.failed)
{
await audioSrc.CloseAudio();
//await testPatternSource.CloseVideo();
}
};
pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) =>
{
bool hasUseCandidate = msg.Attributes.Any(x => x.AttributeType == SIPSorcery.Net.STUNAttributeTypesEnum.UseCandidate);
logger.LogDebug($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
return(Task.FromResult(pc));
}
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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Asterisk + ICE Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
userAgent.OnCallHungup += async(dialog) =>
{
// Give time for the BYE response before exiting.
await Task.Delay(1000);
exitCts.Cancel();
};
var audioExtras = new AudioExtrasSource();
audioExtras.SetSource(AudioSourcesEnum.PinkNoise);
var testPattern = new VideoTestPatternSource(new VpxVideoEncoder());
var pc = new RTCPeerConnection(null);
pc.OnAudioFormatsNegotiated += (formats) => audioExtras.SetAudioSourceFormat(formats.First());
pc.OnVideoFormatsNegotiated += (formats) => testPattern.SetVideoSourceFormat(formats.First());
var audioTrack = new MediaStreamTrack(audioExtras.GetAudioSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
audioExtras.OnAudioSourceEncodedSample += pc.SendAudio;
var videoTrack = new MediaStreamTrack(testPattern.GetVideoSourceFormats(), MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
testPattern.OnVideoSourceEncodedSample += pc.SendVideo;
// Diagnostics.
pc.OnReceiveReport += (re, media, rr) => Console.WriteLine($"RTCP Receive for {media} from {re}\n{rr.GetDebugSummary()}");
pc.OnSendReport += (media, sr) => Console.WriteLine($"RTCP Send for {media}\n{sr.GetDebugSummary()}");
pc.GetRtpChannel().OnStunMessageReceived += (msg, ep, isRelay) => Console.WriteLine($"STUN {msg.Header.MessageType} received from {ep}.");
pc.oniceconnectionstatechange += (state) => Console.WriteLine($"ICE connection state change to {state}.");
// ICE connection state handler.
pc.onconnectionstatechange += (state) =>
{
Console.WriteLine($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.connected)
{
audioExtras.StartAudio();
testPattern.StartVideo();
}
else if (state == RTCPeerConnectionState.failed)
{
if (userAgent.IsCallActive)
{
Console.WriteLine("ICE connection failed, hanging up active call.");
userAgent.Hangup();
}
}
};
// Place the call and wait for the result.
var callTask = userAgent.Call(DESTINATION, USERNAME, PASSWORD, pc);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent != null)
{
if (userAgent.IsCalling || userAgent.IsRinging)
{
Console.WriteLine("Cancelling in progress call.");
userAgent.Cancel();
}
else if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up established call.");
userAgent.Hangup();
}
}
;
exitCts.Cancel();
};
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
Console.WriteLine($"Call to {DESTINATION} succeeded.");
exitCts.Token.WaitHandle.WaitOne();
}
else
{
Console.WriteLine($"Call to {DESTINATION} failed.");
}
Console.WriteLine("Exiting...");
if (userAgent?.IsHangingUp == true)
{
Console.WriteLine("Waiting 1s for the call hangup or cancel to complete...");
await Task.Delay(1000);
}
// Clean up.
sipTransport.Shutdown();
}
private static async Task <RTCPeerConnection> CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(new VpxVideoEncoder(), WEBCAM_NAME);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(new FFmpegVideoEncoder(), WEBCAM_NAME);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(WEBCAM_NAME, 1920, 1080, 30);
//winVideoEP.RestrictFormats(x => x.Codec == SIPSorceryMedia.Abstractions.V1.VideoCodecsEnum.H264);
bool initResult = await winVideoEP.InitialiseVideoSourceDevice();
if (!initResult)
{
throw new ApplicationException("Could not initialise video capture device.");
}
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(winVideoEP.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
winVideoEP.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (videoFormats) =>
winVideoEP.SetVideoSourceFormat(videoFormats.First());
pc.OnAudioFormatsNegotiated += (audioFormats) =>
audioSource.SetAudioSourceFormat(audioFormats.First());
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"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);
}
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}.");
};
};
_peerConnection.onsignalingstatechange += () =>
{
if (_peerConnection.signalingState == RTCSignalingState.have_remote_offer)
{
logger.LogDebug("Remote SDP:");
logger.LogDebug(_peerConnection.remoteDescription.sdp.ToString());
}
else if (_peerConnection.signalingState == RTCSignalingState.have_local_offer)
{
logger.LogDebug("Local SDP:");
logger.LogDebug(_peerConnection.localDescription.sdp.ToString());
}
};
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 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.OnAudioFormatsNegotiated += (formats) => audioSource.SetAudioSourceFormat(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));
}