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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
protected override async Task ExecuteAsync(CancellationToken stoppingToken)
{
// Set up media sources.
_maxSource = new FFmpegFileSource(MP4_PATH, true, new AudioEncoder());
_maxSource.Initialise();
_testPatternSource = new VideoTestPatternSource();
_testPatternEncoder = new FFmpegVideoEndPoint();
_testPatternSource.OnVideoSourceRawSample += _testPatternEncoder.ExternalVideoSourceRawSample;
_musicSource = new AudioExtrasSource(new AudioEncoder(),
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
// The same sources are used for all connected peers (broadcast) so the codecs need
// to be restricted to a single well supported option.
_musicSource.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
_maxSource.RestrictCodecs(new List <VideoCodecsEnum> {
VIDEO_CODEC
});
_testPatternEncoder.RestrictCodecs(new List <VideoCodecsEnum> {
VIDEO_CODEC
});
// Start web socket.
_logger.LogInformation("Starting web socket server...");
WebSocketServer webSocketServer = null;
if (!string.IsNullOrWhiteSpace(_certificatePath))
{
if (!File.Exists(_certificatePath))
{
throw new ApplicationException($"Certificate path could not be found {_certificatePath}.");
}
webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
webSocketServer.SslConfiguration.ServerCertificate = new X509Certificate2(_certificatePath);
webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
webSocketServer.SslConfiguration.EnabledSslProtocols = System.Security.Authentication.SslProtocols.Tls12;
}
else
{
webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
}
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = () => CreatePeerConnection(null));
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>($"/{MAX_URL}",
(peer) => peer.CreatePeerConnection = () => CreatePeerConnection(MAX_URL));
webSocketServer.Start();
_logger.LogInformation($"Waiting for web socket connections on {webSocketServer.Address}:{webSocketServer.Port}...");
await Task.Delay(Timeout.Infinite, stoppingToken);
}
/// <summary>
/// An asynchronous task that attempts to initiate a new call to a listening UAS.
/// </summary>
/// <param name="sipTransport">The transport object to use for the send.</param>
/// <param name="dst">The destination end point to send the request to.</param>
/// <returns>True if the expected response was received, false otherwise.</returns>
private static async Task <bool> InitiateCallTaskAsync(SIPTransport sipTransport, SIPURI dst)
{
//UdpClient hepClient = new UdpClient(0, AddressFamily.InterNetwork);
try
{
//sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
//{
// logger.LogDebug($"Request sent: {localEP}->{remoteEP}");
// logger.LogDebug(req.ToString());
// //var hepBuffer = HepPacket.GetBytes(localEP, remoteEP, DateTimeOffset.Now, 333, "myHep", req.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
//sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
//{
// logger.LogDebug($"Response received: {localEP}<-{remoteEP}");
// logger.LogDebug(resp.ToString());
// //var hepBuffer = HepPacket.GetBytes(remoteEP, localEP, DateTimeOffset.Now, 333, "myHep", resp.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
var ua = new SIPUserAgent(sipTransport, null);
ua.ClientCallTrying += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallRinging += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallFailed += (uac, err, resp) => logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
ua.ClientCallAnswered += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
var result = await ua.Call(dst.ToString(), null, null, voipMediaSession);
ua.Hangup();
await Task.Delay(200);
return(result);
}
catch (Exception excp)
{
logger.LogError($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
private Task <SIPSorcery.Net.RTCPeerConnection> CreatePeerConnection()
{
var pc = new SIPSorcery.Net.RTCPeerConnection(null);
// Set up sources and hook up send events to peer connection.
AudioExtrasSource audioSrc = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
});
audioSrc.OnAudioSourceEncodedSample += pc.SendAudio;
//var testPatternSource = new VideoTestPatternSource();
//testPatternSource.SetMaxFrameRate(true);
//testPatternSource.OnVideoSourceRawSample += VideoEncoderEndPoint.ExternalVideoSourceRawSample;
VideoEncoderEndPoint.OnVideoSourceEncodedSample += pc.SendVideo;
// Add tracks.
var audioTrack = new SIPSorcery.Net.MediaStreamTrack(audioSrc.GetAudioSourceFormats(), SIPSorcery.Net.MediaStreamStatusEnum.SendOnly);
pc.addTrack(audioTrack);
var videoTrack = new SIPSorcery.Net.MediaStreamTrack(VideoEncoderEndPoint.GetVideoSourceFormats(), SIPSorcery.Net.MediaStreamStatusEnum.SendOnly);
pc.addTrack(videoTrack);
// Handlers to set the codecs to use on the sources once the SDP negotiation is complete.
pc.OnVideoFormatsNegotiated += (sdpFormat) => VideoEncoderEndPoint.SetVideoSourceFormat(SIPSorcery.Net.SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.OnAudioFormatsNegotiated += (sdpFormat) =>
audioSrc.SetAudioSourceFormat(SIPSorcery.Net.SDPMediaFormatInfo.GetAudioCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.OnTimeout += (mediaType) => logger.LogDebug($"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($"STUN {msg.Header.MessageType} received from {ep}, use candidate {hasUseCandidate}.");
};
return(Task.FromResult(pc));
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Convert Audio");
Log = AddConsoleLogger();
//WaveFormatConversionStream converter = new WaveFormatConversionStream(_format_s16le48k, )
_waveFile = new WaveFileWriter("output_s16le48k.mp3", _format_s16le48k);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) =>
{
Console.WriteLine("Call hungup.");
_waveFile?.Close();
};
//EnableTraceLogs(sipTransport);
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictFormats((format) => format.Codec == AudioCodecsEnum.PCMU);
var rtpSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpSession.OnAudioFormatsNegotiated += (formats) =>
{
_ratio = (double)(OUT_SAMPLE_RATE / formats.First().RtpClockRate);
};
rtpSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
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 VoIPMediaSession CreateMockVoIPMediaEndPoint(Func <AudioFormat, bool> audioFormatFilter = null)
{
var audioSource = new AudioExtrasSource();
audioSource.RestrictFormats(audioFormatFilter);
MediaEndPoints mockEndPoints = new MediaEndPoints
{
AudioSource = audioSource
};
return(new VoIPMediaSession(mockEndPoints));
}
private VoIPMediaSession CreateMockVoIPMediaEndPoint(List <AudioCodecsEnum> supportedCodecs = null)
{
var audioSource = new AudioExtrasSource();
audioSource.RestrictCodecs(supportedCodecs);
MediaEndPoints mockEndPoints = new MediaEndPoints
{
AudioSource = audioSource
};
return(new VoIPMediaSession(mockEndPoints));
}
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($"ICE connection state change to {state}.");
return(Task.FromResult(pc));
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <AudioCodecsEnum> codecs = new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA, AudioCodecsEnum.G722
};
var audioSource = AudioSourcesEnum.SineWave;
if (string.IsNullOrEmpty(dst) || !Enum.IsDefined(typeof(AudioSourcesEnum), dst) || !Enum.TryParse(dst, out audioSource))
{
audioSource = AudioSourcesEnum.Music;
}
Log.LogInformation($"RTP audio session source set to {audioSource}.");
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = audioSource
});
audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec));
var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpAudioSession.AcceptRtpFromAny = true;
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}
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($"ICE connection state change to {state}.");
return(pc);
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
Log = AddConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2(SIPS_CERTIFICATE_PATH);
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
VoIPMediaSession rtpSession = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
rtpSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = extrasSource
});
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
await sipTransport.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
rtpCts = new CancellationTokenSource();
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.Close(null);
};
rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
await rtpSession.Start();
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Log.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.Close(null);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpSession?.Close(null);
rtpCts?.Cancel();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpSession?.Close(null);
rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
Log.LogInformation("Quitting...");
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
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));
}
/// <summary>
/// An asynchronous task that attempts to initiate a new call to a listening UAS.
/// </summary>
/// <param name="sipTransport">The transport object to use for the send.</param>
/// <param name="dst">The destination end point to send the request to.</param>
/// <returns>True if the expected response was received, false otherwise.</returns>
private static async Task <bool> InitiateCallTaskAsync(SIPTransport sipTransport, SIPURI dst, Scenarios scenario)
{
//UdpClient hepClient = new UdpClient(0, AddressFamily.InterNetwork);
try
{
//sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
//{
// logger.LogDebug($"Request sent: {localEP}->{remoteEP}");
// logger.LogDebug(req.ToString());
// //var hepBuffer = HepPacket.GetBytes(localEP, remoteEP, DateTimeOffset.Now, 333, "myHep", req.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
//sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
//{
// logger.LogDebug($"Response received: {localEP}<-{remoteEP}");
// logger.LogDebug(resp.ToString());
// //var hepBuffer = HepPacket.GetBytes(remoteEP, localEP, DateTimeOffset.Now, 333, "myHep", resp.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
var ua = new SIPUserAgent(sipTransport, null);
ua.ClientCallTrying += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallRinging += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallFailed += (uac, err, resp) => logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
ua.ClientCallAnswered += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
var result = await ua.Call(dst.ToString(), null, null, voipMediaSession);
if (scenario == Scenarios.uacw)
{
// Wait for the remote party to hangup the call.
logger.LogDebug("Waiting for the remote party to hangup the call...");
ManualResetEvent hangupMre = new ManualResetEvent(false);
ua.OnCallHungup += (dialog) => hangupMre.Set();
hangupMre.WaitOne();
logger.LogDebug("Call hungup by remote party.");
}
else
{
// We hangup the call after 1s.
await Task.Delay(1000);
ua.Hangup();
}
// Need a bit of time for the BYE to complete.
await Task.Delay(500);
return(result);
}
catch (Exception excp)
{
logger.LogError($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
private static RTCPeerConnection CreatePeerConnection()
{
RTCConfiguration config = new RTCConfiguration
{
iceServers = new List <RTCIceServer> {
new RTCIceServer {
urls = STUN_URL
}
}
};
var pc = new RTCPeerConnection(config);
WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint();
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(false, 640, 480, 30);
//WindowsVideoEndPoint winVideoEP = new WindowsVideoEndPoint(false, 1920, 1080, 30);
//await winVideoEP.InitialiseVideoSourceDevice();
var audioSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
MediaStreamTrack videoTrack = new MediaStreamTrack(winVideoEP.GetVideoSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(videoTrack);
MediaStreamTrack audioTrack = new MediaStreamTrack(audioSource.GetAudioSourceFormats(), MediaStreamStatusEnum.SendRecv);
pc.addTrack(audioTrack);
winVideoEP.OnVideoSourceEncodedSample += pc.SendVideo;
audioSource.OnAudioSourceEncodedSample += pc.SendAudio;
pc.OnVideoFormatsNegotiated += (sdpFormat) =>
winVideoEP.SetVideoSourceFormat(SDPMediaFormatInfo.GetVideoCodecForSdpFormat(sdpFormat.First().FormatCodec));
pc.onconnectionstatechange += async(state) =>
{
logger.LogDebug($"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);
}
