public void AckRecognitionUnitTest()
{
SIPTransport clientTransport = null;
SIPTransport serverTransport = null;
try
{
SIPTransactionEngine clientEngine = new SIPTransactionEngine(); // Client side of the INVITE.
clientTransport = new SIPTransport(MockSIPDNSManager.Resolve, clientEngine, new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)), false);
SetTransportTraceEvents(clientTransport);
SIPTransactionEngine serverEngine = new SIPTransactionEngine(); // Server side of the INVITE.
UASInviteTransaction serverTransaction = null;
serverTransport = new SIPTransport(MockSIPDNSManager.Resolve, serverEngine, new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)), false);
SetTransportTraceEvents(serverTransport);
serverTransport.SIPTransportRequestReceived += (localEndPoint, remoteEndPoint, sipRequest) =>
{
logger.LogDebug("Server Transport Request In: " + sipRequest.Method + ".");
serverTransaction = serverTransport.CreateUASTransaction(sipRequest, null);
SetTransactionTraceEvents(serverTransaction);
serverTransaction.GotRequest(localEndPoint, remoteEndPoint, sipRequest);
};
SIPURI dummyURI = SIPURI.ParseSIPURI("sip:dummy@" + serverTransport.GetSIPChannels().First().GetLocalSIPEndPointForDestination(IPAddress.Loopback).GetIPEndPoint());
SIPRequest inviteRequest = GetDummyINVITERequest(dummyURI);
// Send the invite to the server side.
UACInviteTransaction clientTransaction = new UACInviteTransaction(clientTransport, inviteRequest, null);
SetTransactionTraceEvents(clientTransaction);
clientEngine.AddTransaction(clientTransaction);
clientTransaction.SendInviteRequest(inviteRequest);
Thread.Sleep(500);
Assert.True(clientTransaction.TransactionState == SIPTransactionStatesEnum.Completed, "Client transaction in incorrect state.");
Assert.True(serverTransaction.TransactionState == SIPTransactionStatesEnum.Confirmed, "Server transaction in incorrect state.");
}
finally
{
if (clientTransport != null)
{
clientTransport.Shutdown();
}
if (serverTransport != null)
{
serverTransport.Shutdown();
}
}
}
public Task StartAsync(CancellationToken cancellationToken)
{
Logger.LogDebug("SIP hosted service starting...");
int listenPort = Configuration.GetValue <int>("SIPListenPort", DEFAULT_SIP_LISTEN_PORT);
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, listenPort)));
var listeningEP = _sipTransport.GetSIPChannels().First().ListeningSIPEndPoint;
Logger.LogInformation($"SIP transport listening on {listeningEP}.");
EnableTraceLogs(_sipTransport, true);
_bindingsManager.Start();
_registrarCore.Start(REGISTRAR_CORE_WORKER_THREADS);
_b2bUserAgentCore.Start(B2BUA_CORE_WORKER_THREADS);
_sipTransport.SIPTransportRequestReceived += OnRequest;
return(Task.CompletedTask);
}
public async Task AttendedTransfereeUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// User agents A and B can use the same transport as they don't auto-answer incoming calls.
SIPTransport sipTransportCaller = new SIPTransport();
sipTransportCaller.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var userAgentA = new SIPUserAgent(sipTransportCaller, null);
var userAgentB = new SIPUserAgent(sipTransportCaller, null);
SIPTransport sipTransportC = new SIPTransport();
sipTransportC.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var userAgentC = new SIPUserAgent(sipTransportC, null);
SIPTransport sipTransportD = new SIPTransport();
sipTransportD.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var userAgentD = new SIPUserAgent(sipTransportD, null);
logger.LogDebug($"sip transport for UA's A and B listening on: {sipTransportCaller.GetSIPChannels()[0].ListeningSIPEndPoint}.");
logger.LogDebug($"sip transport for UA C listening on: {sipTransportC.GetSIPChannels()[0].ListeningSIPEndPoint}.");
logger.LogDebug($"sip transport for UA D listening on: {sipTransportD.GetSIPChannels()[0].ListeningSIPEndPoint}.");
// Set up auto-answer for UA's C and D:
foreach (var userAgent in new List <SIPUserAgent> {
userAgentC, userAgentD
})
{
userAgent.ServerCallCancelled += (uas) => logger.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => logger.LogDebug("Call hungup by remote party.");
userAgent.OnIncomingCall += async(ua, req) =>
{
var uas = ua.AcceptCall(req);
bool answerResult = await ua.Answer(uas, CreateMediaSession());
logger.LogDebug($"Answer incoming call result {answerResult}.");
};
}
// Place the two calls from A to C and B to D.
var dstUriC = sipTransportC.GetSIPChannels()[0].GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"UA-A attempting call UA-C on {dstUriC}.");
var callResultAtoC = await userAgentA.Call(dstUriC.ToString(), null, null, CreateMediaSession());
logger.LogDebug($"Client agent answer result for A to C {callResultAtoC}.");
Assert.True(callResultAtoC);
var dstUriD = sipTransportD.GetSIPChannels()[0].GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"UA-B attempting call UA-D on {dstUriD}.");
var callResultBtoD = await userAgentB.Call(dstUriD.ToString(), null, null, CreateMediaSession());
logger.LogDebug($"Client agent answer result for B to D {callResultBtoD}.");
Assert.True(callResultBtoD);
Assert.True(userAgentA.IsCallActive);
Assert.True(userAgentB.IsCallActive);
Assert.True(userAgentC.IsCallActive);
Assert.True(userAgentD.IsCallActive);
// Initiate attended transfer. A sends REFER request to C such that:
// - A sends a REFER request to C,
// - The REFER request Refer-To header tells C who to call and what to put in its INVITE request Replaces header,
// - The INVITE from C to D tells D this new call from C replaces its call with B,
// - When D answers it hangs up its call with B,
// - When C gets the Ok response from C it hangs up its call with A.
CancellationTokenSource cts = new CancellationTokenSource();
bool transferResult = await userAgentA.AttendedTransfer(userAgentB.Dialogue, TimeSpan.FromSeconds(2), cts.Token);
// This means the REFER request was accepted but the transfer still needs to be actioned.
Assert.True(transferResult);
// Give the transfer time to be processed.
await Task.Delay(2000);
Assert.False(userAgentA.IsCallActive);
Assert.False(userAgentB.IsCallActive);
Assert.True(userAgentC.IsCallActive);
Assert.True(userAgentD.IsCallActive);
sipTransportCaller.Shutdown();
sipTransportC.Shutdown();
sipTransportD.Shutdown();
}
public async Task HandleInvalidSdpPortOnPlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// This transport will act as the call receiver. It allows the test to provide a
// tailored response to an incoming call.
SIPTransport calleeTransport = new SIPTransport();
// This transport will be used by the SIPUserAgent being tested to place the call.
SIPTransport callerTransport = new SIPTransport();
RTPSession rtpSession = new RTPSession(false, false, false);
try
{
calleeTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
calleeTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
if (req.Method != SIPMethodsEnum.INVITE)
{
SIPResponse notAllowedResponse = SIPResponse.GetResponse(req, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await calleeTransport.SendResponseAsync(notAllowedResponse);
}
else
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(calleeTransport, req, null);
var uas = new SIPServerUserAgent(calleeTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = @"
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp, null, SIPDialogueTransferModesEnum.NotAllowed);
}
};
SIPUserAgent userAgent = new SIPUserAgent(callerTransport, null);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
SIPURI dstUri = new SIPURI(SIPSchemesEnum.sip, calleeTransport.GetSIPChannels().First().ListeningSIPEndPoint);
var result = await userAgent.Call(dstUri.ToString(), null, null, rtpSession);
Assert.False(result);
}
finally
{
rtpSession?.Close("normal");
callerTransport?.Shutdown();
calleeTransport?.Shutdown();
}
}
static async Task Main(string[] args)
{
Console.WriteLine("SIPSorcery Video Phone Command Line Demo");
Console.WriteLine("Press ctrl-c to exit.");
Log = AddConsoleLogger();
ManualResetEvent exitMRE = new ManualResetEvent(false);
ManualResetEvent waitForCallMre = new ManualResetEvent(false);
var parseResult = Parser.Default.ParseArguments <Options>(args);
_options = (parseResult as Parsed <Options>)?.Value;
if (parseResult.Tag != ParserResultType.NotParsed)
{
if (_options.ListCameras)
{
#region List webcams.
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
}
else
{
var index = 0;
foreach (var webcam in webcams)
{
Console.WriteLine($"{index}: \"{webcam.Name}\", use --cam={index}.");
index++;
}
}
#endregion
}
else if (_options.ListFormats != null)
{
#region List webcam formats.
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
}
else if (_options.ListFormats >= webcams.Count)
{
Console.WriteLine($"No webcam available for index {_options.ListFormats}.");
}
else
{
string webcamName = webcams[_options.ListFormats.Value].Name;
var formats = await WindowsVideoEndPoint.GetDeviceFrameFormats(webcamName);
Console.WriteLine($"Video frame formats for {webcamName}.");
foreach (var vidFmt in formats)
{
float vidFps = vidFmt.MediaFrameFormat.FrameRate.Numerator / vidFmt.MediaFrameFormat.FrameRate.Denominator;
string pixFmt = vidFmt.MediaFrameFormat.Subtype == WindowsVideoEndPoint.MF_I420_PIXEL_FORMAT ? "I420" : vidFmt.MediaFrameFormat.Subtype;
Console.WriteLine($"{vidFmt.Width}x{vidFmt.Height} {vidFps:0.##}fps {pixFmt}");
}
}
#endregion
}
else
{
string webcamName = null;
if (_options.WebcamIndex != null)
{
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
Application.Exit();
}
else if (webcams.Count < _options.WebcamIndex)
{
Console.WriteLine($"No webcam available for index {_options.WebcamIndex}.");
Application.Exit();
}
else
{
webcamName = webcams[_options.WebcamIndex.Value].Name;
Console.WriteLine($"Using webcam {webcamName}.");
}
}
_sipTransport = new SIPTransport();
if (string.IsNullOrEmpty(_options.CallDestination))
{
// We haven't been asked to place a call so we're listening.
IPAddress listenAddress = (System.Net.Sockets.Socket.OSSupportsIPv6) ? IPAddress.IPv6Any : IPAddress.Any;
var listenEndPoint = new IPEndPoint(listenAddress, SIP_PORT_DEFAULT);
try
{
SIPUDPChannel udpChannel = new SIPUDPChannel(listenEndPoint, true);
_sipTransport.AddSIPChannel(udpChannel);
}
catch (ApplicationException appExcp)
{
Console.WriteLine($"Failed to create UDP SIP channel on {listenEndPoint}, error {appExcp.Message}.");
SIPUDPChannel udpChannel = new SIPUDPChannel(new IPEndPoint(listenAddress, 0), true);
_sipTransport.AddSIPChannel(udpChannel);
}
var listeningEP = _sipTransport.GetSIPChannels().First().ListeningSIPEndPoint;
Console.WriteLine($"Listening for incoming call on {listeningEP}.");
}
EnableTraceLogs(_sipTransport);
// Open a window to display the video feed from the remote SIP party.
_form = new Form();
_form.Text = string.IsNullOrEmpty(_options.CallDestination) ? "Listener" : "Caller";
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_localVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, 0),
Visible = true
};
_remoteVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, VIDEO_FRAME_HEIGHT),
Visible = true
};
_form.Controls.Add(_localVideoPicBox);
_form.Controls.Add(_remoteVideoPicBox);
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.OnCallHungup += (dialog) => exitMRE.Set();
WindowsAudioEndPoint windowsAudioEndPoint = null;
if (!_options.NoAudio)
{
windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudioEndPoint.RestrictFormats(x => x.Codec == AudioCodecsEnum.G722);
}
MediaEndPoints mediaEndPoints = null;
if (_options.TestPattern && _options.WebcamIndex == null)
{
var testPattern = new VideoTestPatternSource(new FFmpegVideoEncoder());
var decoderSink = new DecoderVideoSink(new FFmpegVideoEncoder());
//var decoderSink = new DecoderVideoSink(new VpxVideoEncoder());
testPattern.RestrictFormats(format => format.Codec == VIDEO_CODEC);
decoderSink.RestrictFormats(format => format.Codec == VIDEO_CODEC);
mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = decoderSink,
VideoSource = testPattern,
};
}
else
{
WindowsVideoEndPoint windowsVideoEndPoint = webcamName switch
{
null => new WindowsVideoEndPoint(new FFmpegVideoEncoder()),
_ => new WindowsVideoEndPoint(new FFmpegVideoEncoder(), webcamName),
};
windowsVideoEndPoint.RestrictFormats(format => format.Codec == VIDEO_CODEC);
mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
}
mediaEndPoints.VideoSource.OnVideoSourceRawSample += (uint durationMilliseconds, int width, int height, byte[] sample, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
int stride = width * 3;
if (pixelFormat == VideoPixelFormatsEnum.I420)
{
sample = PixelConverter.I420toBGR(sample, width, height, out stride);
}
if (_localVideoPicBox.Width != width || _localVideoPicBox.Height != height)
{
Log.LogDebug($"Adjusting local video display from {_localVideoPicBox.Width}x{_localVideoPicBox.Height} to {width}x{height}.");
_localVideoPicBox.Width = width;
_localVideoPicBox.Height = height;
}
unsafe
{
fixed(byte *s = sample)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap(width, height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_localVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
waitForCallMre.Set();
exitMRE.Set();
};
if (string.IsNullOrEmpty(_options.CallDestination))
{
ActivateForm();
userAgent.OnIncomingCall += async(ua, req) =>
{
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
if (voipMediaSession.VideoLocalTrack != null)
{
voipMediaSession.VideoLocalTrack.MaximumBandwidth = MAXIMUM_VIDEO_BANDWIDTH;
}
var uas = userAgent.AcceptCall(req);
await userAgent.Answer(uas, voipMediaSession);
Console.WriteLine("Starting local video source...");
await mediaEndPoints.VideoSource.StartVideo().ConfigureAwait(false);
waitForCallMre.Set();
};
Console.WriteLine("Waiting for incoming call...");
waitForCallMre.WaitOne();
}
else
{
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
if (voipMediaSession.VideoLocalTrack != null)
{
voipMediaSession.VideoLocalTrack.MaximumBandwidth = MAXIMUM_VIDEO_BANDWIDTH;
}
ActivateForm();
Console.WriteLine("Starting local video source...");
await mediaEndPoints.VideoSource.StartVideo().ConfigureAwait(false);
// Place the call and wait for the result.
Task <bool> callTask = userAgent.Call(_options.CallDestination, null, null, voipMediaSession);
callTask.Wait(CALL_TIMEOUT_SECONDS * 1000);
}
if (userAgent.IsCallActive)
{
Log.LogInformation("Call attempt successful.");
mediaEndPoints.VideoSink.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
unsafe
{
if (_remoteVideoPicBox.Width != (int)width || _remoteVideoPicBox.Height != (int)height)
{
Log.LogDebug($"Adjusting remote video display from {_remoteVideoPicBox.Width}x{_remoteVideoPicBox.Height} to {width}x{height}.");
_remoteVideoPicBox.Width = (int)width;
_remoteVideoPicBox.Height = (int)height;
}
fixed(byte *s = bmp)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap((int)width, (int)height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_remoteVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
}
else
{
Log.LogWarning("Call attempt failed.");
Console.WriteLine("Press ctrl-c to exit.");
}
exitMRE.WaitOne();
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up.");
userAgent.Hangup();
}
Task.Delay(1000).Wait();
// Clean up.
if (_form.Handle != IntPtr.Zero)
{
_form.BeginInvoke(new Action(() => _form.Close()));
}
_sipTransport.Shutdown();
}
}
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
winAudio.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogInformation($"Incoming call from {req.Header.From.FriendlyDescription()} at {req.RemoteSIPEndPoint}.");
var uas = userAgent.AcceptCall(req);
if (userAgent?.IsCallActive == true)
{
// If we are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request.");
uas.Reject(SIPResponseStatusCodesEnum.BusyHere, null);
}
else
{
var voipSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipSession.AcceptRtpFromAny = true;
var answerResult = await userAgent.Answer(uas, voipSession);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
var voipSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipSession.AcceptRtpFromAny = true;
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, voipSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
(userAgent.MediaSession as VoIPMediaSession).TakeOffHold();
userAgent.TakeOffHold();
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
await(userAgent.MediaSession as VoIPMediaSession).PutOnHold();
userAgent.PutOnHold();
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
_currentDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
RtpAVSession rtpAVSession = null;
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (userAgent?.IsCallActive == true)
{
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
// If we are already on a call return a busy response.
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
else
{
Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await userAgent.Answer(incomingCall, rtpAVSession);
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, rtpAVSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
userAgent.TakeOffHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
userAgent.PutOnHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions
{
AudioSource = AudioSourcesEnum.Music,
SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>
{
{ SDPMediaFormatsEnum.PCMU, _currentDir + "/" + AUDIO_FILE_PCMU }
}
}, null);
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpAVSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main()
{
Console.WriteLine("SIPSorcery SIP to WebRTC example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
//EnableTraceLogs(sipTransport);
// Start web socket.
Console.WriteLine("Starting web socket server...");
var webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT);
webSocketServer.AddWebSocketService <WebRTCWebSocketPeer>("/", (peer) => peer.CreatePeerConnection = CreatePeerConnection);
webSocketServer.Start();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null, true);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogInformation($"Incoming call request from {req.RemoteSIPEndPoint}: {req.StatusLine}.");
var incomingCall = userAgent.AcceptCall(req);
var rtpSession = new RTPSession(false, false, false);
rtpSession.AcceptRtpFromAny = true;
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false,
new List <SDPAudioVideoMediaFormat> {
new SDPAudioVideoMediaFormat(SDPWellKnownMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
await userAgent.Answer(incomingCall, rtpSession);
rtpSession.OnRtpPacketReceived += ForwardMediaToPeerConnection;
Log.LogInformation($"Answered incoming call from {req.Header.From.FriendlyDescription()} at {req.RemoteSIPEndPoint}.");
_rtpSession = rtpSession;
};
Console.WriteLine($"Waiting for browser web socket connection to {webSocketServer.Address}:{webSocketServer.Port}...");
var contactURI = new SIPURI(SIPSchemesEnum.sip, sipTransport.GetSIPChannels().First().ListeningSIPEndPoint);
Console.WriteLine($"Waiting for incoming SIP call to {contactURI}.");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
_rtpSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
public void AckRecognitionUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport clientTransport = null;
SIPTransport serverTransport = null;
try
{
TaskCompletionSource <bool> uasConfirmedTask = new TaskCompletionSource <bool>(TaskCreationOptions.RunContinuationsAsynchronously);
// Client side of the call.
clientTransport = new SIPTransport();
clientTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Loopback, 0)));
var clientEngine = clientTransport.m_transactionEngine;
SetTransportTraceEvents(clientTransport);
// Server side of the call.
UASInviteTransaction serverTransaction = null;
serverTransport = new SIPTransport();
serverTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Loopback, 0)));
SIPTransactionEngine serverEngine = serverTransport.m_transactionEngine;
SetTransportTraceEvents(serverTransport);
serverTransport.SIPTransportRequestReceived += (localEndPoint, remoteEndPoint, sipRequest) =>
{
logger.LogDebug("Server Transport Request In: " + sipRequest.Method + ".");
serverTransaction = new UASInviteTransaction(serverTransport, sipRequest, null);
SetTransactionTraceEvents(serverTransaction);
serverTransaction.NewCallReceived += (lep, rep, sipTransaction, newCallRequest) =>
{
logger.LogDebug("Server new call received.");
var busyResponse = SIPResponse.GetResponse(newCallRequest, SIPResponseStatusCodesEnum.BusyHere, null);
(sipTransaction as UASInviteTransaction).SendFinalResponse(busyResponse);
return(Task.FromResult(SocketError.Success));
};
serverTransaction.TransactionStateChanged += (tx) =>
{
if (tx.TransactionState == SIPTransactionStatesEnum.Confirmed)
{
if (!uasConfirmedTask.TrySetResult(true))
{
logger.LogWarning($"AckRecognitionUnitTest: FAILED to set result on CompletionSource.");
}
}
};
serverTransaction.GotRequest(localEndPoint, remoteEndPoint, sipRequest);
return(Task.FromResult(0));
};
SIPURI dummyURI = new SIPURI("dummy", serverTransport.GetSIPChannels().First().ListeningEndPoint.ToString(), null, SIPSchemesEnum.sip);
SIPRequest inviteRequest = GetDummyINVITERequest(dummyURI);
// Send the invite to the server side.
UACInviteTransaction clientTransaction = new UACInviteTransaction(clientTransport, inviteRequest, null);
SetTransactionTraceEvents(clientTransaction);
clientEngine.AddTransaction(clientTransaction);
clientTransaction.SendInviteRequest();
if (!uasConfirmedTask.Task.Wait(TRANSACTION_EXCHANGE_TIMEOUT_MS))
{
logger.LogWarning($"Tasks timed out");
}
Assert.True(clientTransaction.TransactionState == SIPTransactionStatesEnum.Confirmed, "Client transaction in incorrect state.");
Assert.True(serverTransaction.TransactionState == SIPTransactionStatesEnum.Confirmed, "Server transaction in incorrect state.");
}
finally
{
clientTransport.Shutdown();
serverTransport.Shutdown();
}
}
public Task StartAsync(CancellationToken cancellationToken)
{
_logger.LogDebug("SIP hosted service starting...");
// Get application configuration settings.
int listenPort = _config.GetValue <int>(ConfigKeys.SIP_LISTEN_PORT, DEFAULT_SIP_LISTEN_PORT);
int tlsListenPort = _config.GetValue <int>(ConfigKeys.SIP_TLS_LISTEN_PORT, DEFAULT_SIPS_LISTEN_PORT);
// Set the fields used for setting the SIP Contact header.
_publicContactHostname = _config.GetValue <string>(ConfigKeys.SIP_PUBLIC_CONTACT_HOSTNAME, null);
IPAddress.TryParse(_config.GetValue <string>(ConfigKeys.SIP_PUBLIC_CONTACT_PUBLICIPV4, null), out _publicContactIPv4);
IPAddress.TryParse(_config.GetValue <string>(ConfigKeys.SIP_PUBLIC_CONTACT_PUBLICIPV6, null), out _publicContactIPv6);
IConfigurationSection sipPrivateSubnets = _config.GetSection(ConfigKeys.SIP_PRIVATE_CONTACT_SUBNETS);
_logger.LogInformation($"SIP transport public contact hostname set to {_publicContactHostname}.");
_logger.LogInformation($"SIP transport public contact IPv4 set to {_publicContactIPv4}.");
_logger.LogInformation($"SIP transport public contact IPv6 set to {_publicContactIPv6}.");
if (sipPrivateSubnets != null)
{
List <Func <IPAddress, bool> > isInSubnetFunctions = new List <Func <IPAddress, bool> >();
foreach (string subnet in sipPrivateSubnets.Get <string[]>())
{
_logger.LogInformation($"SIP transport private subnet {subnet}.");
if (IPNetwork.TryParse(subnet, out var network))
{
isInSubnetFunctions.Add((ipaddr) =>
ipaddr.AddressFamily == network.AddressFamily &&
network.Contains(ipaddr));
}
else
{
_logger.LogWarning($"IP network not recognised {subnet}.");
}
}
if (isInSubnetFunctions.Count > 0)
{
_isPrivateSubnet = (ipaddr) =>
{
foreach (var isInFunc in isInSubnetFunctions)
{
if (isInFunc(ipaddr))
{
return(true);
}
}
return(false);
};
}
_sipFail2Ban.IsPrivateSubnet = _isPrivateSubnet;
}
// Create SIP channels.
if (Socket.OSSupportsIPv6)
{
if (_tlsCertificate != null)
{
_sipTransport.AddSIPChannel(new SIPTLSChannel(_tlsCertificate, new IPEndPoint(IPAddress.IPv6Any, tlsListenPort), true));
}
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, listenPort), true));
_sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.IPv6Any, listenPort), true));
}
else
{
if (_tlsCertificate != null)
{
_sipTransport.AddSIPChannel(new SIPTLSChannel(_tlsCertificate, new IPEndPoint(IPAddress.Any, tlsListenPort)));
}
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, listenPort)));
_sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, listenPort)));
}
var listeningEP = _sipTransport.GetSIPChannels().First().ListeningSIPEndPoint;
_logger.LogInformation($"SIP transport listening on {listeningEP}.");
_sipTransport.CustomiseRequestHeader = CustomiseContact;
_sipTransport.CustomiseResponseHeader = CustomiseContact;
_bindingsManager.Start();
_registrarCore.Start(REGISTRAR_CORE_WORKER_THREADS);
_b2bUserAgentCore.Start(B2BUA_CORE_WORKER_THREADS);
_subscriberCore.Start(SUBSCRIBER_CORE_WORKER_THREADS);
_sipTransport.SIPTransportRequestReceived += OnRequest;
// Warm up the dialplan so it's ready for the first call.
_ = Task.Run(async() =>
{
var dp = await _sipDialPlanManager.LoadDialPlan();
if (dp != null)
{
_sipDialPlanManager.CompileDialPlan(dp.DialPlanScript, DateTime.Parse(dp.LastUpdate));
}
});
return(Task.CompletedTask);
}
