static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started PortAudio Demo (YMMV)");
AddConsoleLogger();
var userAgent = new SIPUserAgent();
var portAudioEndPoint = new PortAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(portAudioEndPoint.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
await Task.Delay(1000);
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new WindowsAudioRtpSession();
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
Task.Delay(1000).Wait();
}
// Clean up.
sipTransport.Shutdown();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSink = winAudio, AudioSource = winAudio
});
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, null);
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"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();
SIPSorcery.Net.DNSManager.Stop();
}
public async Task CancelCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport aliceTransport = new SIPTransport();
aliceTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var alice = new SIPUserAgent(aliceTransport, null, true);
SIPServerUserAgent uas = null;
// Auto accept but NOT answering.
alice.OnIncomingCall += (ua, req) => uas = ua.AcceptCall(req);
SIPTransport bobTransport = new SIPTransport();
bobTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var bob = new SIPUserAgent(bobTransport, null, true);
var callTask = bob.Call(alice.ContactURI.ToString(), null, null, CreateMediaSession());
await Task.Delay(500);
Assert.True(bob.IsRinging);
Assert.NotNull(uas);
Assert.False(uas.IsCancelled);
bob.Cancel();
await Task.Delay(500);
Assert.False(alice.IsCallActive);
Assert.False(bob.IsCallActive);
Assert.True(uas.IsCancelled);
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, err, resp) =>
{
Console.WriteLine($"Call failed {err}");
_waveFile?.Close();
};
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession voipSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
voipSession.OnRtpPacketReceived += OnRtpPacketReceived;
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
else
{
Console.WriteLine("Cancelling call");
userAgent.Cancel();
}
};
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipSession);
Console.WriteLine($"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();
}
/// <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($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Convert Audio");
Log = AddConsoleLogger();
//WaveFormatConversionStream converter = new WaveFormatConversionStream(_format_s16le48k, )
_waveFile = new WaveFileWriter("output_s16le48k.mp3", _format_s16le48k);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) =>
{
Console.WriteLine("Call hungup.");
_waveFile?.Close();
};
//EnableTraceLogs(sipTransport);
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictFormats((format) => format.Codec == AudioCodecsEnum.PCMU);
var rtpSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpSession.OnAudioFormatsNegotiated += (formats) =>
{
_ratio = (double)(OUT_SAMPLE_RATE / formats.First().RtpClockRate);
};
rtpSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"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();
}
public async Task PlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport serverTransport = new SIPTransport();
SIPUDPChannel udpChannel = new SIPUDPChannel(IPAddress.Loopback, 0);
serverTransport.AddSIPChannel(udpChannel);
// Set up two user agents: one to answer the test call and one to place it.
SIPUserAgent userAgentServer = new SIPUserAgent(serverTransport, null);
SIPUserAgent userAgentClient = new SIPUserAgent(new SIPTransport(), null);
serverTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
logger.LogDebug("Request received: " + req.StatusLine);
var uas = userAgentServer.AcceptCall(req);
RtpAudioSession serverAudioSession = new RtpAudioSession(
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
},
new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
});
var answerResult = await userAgentServer.Answer(uas, serverAudioSession);
logger.LogDebug($"Server agent answer result {answerResult}.");
Assert.True(answerResult);
};
var dstUri = udpChannel.GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"Attempting call to {dstUri.ToString()}.");
RtpAudioSession clientAudioSession = new RtpAudioSession(
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
},
new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
});
var callResult = await userAgentClient.Call(dstUri.ToString(), null, null, clientAudioSession);
logger.LogDebug($"Client agent answer result {callResult }.");
Assert.True(callResult);
Assert.Equal(SIPDialogueStateEnum.Confirmed, userAgentClient.Dialogue.DialogueState);
Assert.Equal(SIPDialogueStateEnum.Confirmed, userAgentServer.Dialogue.DialogueState);
}
static async Task DialNumber()
{
string fromHeader = (new SIPFromHeader(USERNAME, new SIPURI(USERNAME, DOMAIN, null), null)).ToString();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(USERNAME, PASSWORD, DEFAULT_CALL_DESTINATION, fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, null, null);
callDescriptor.CallId = "12028883999";
userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallRinging += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Failed: {err}, Status code: {resp?.StatusCode}");
userAgent.ClientCallAnswered += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.OnDtmfTone += (key, duration) => OnDtmfTone(userAgent, key, duration);
userAgent.OnRtpEvent += (evt, hdr) => Console.WriteLine($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
userAgent.OnCallHungup += OnHangup;
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var callResult = await userAgent.Call(callDescriptor, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
if (callResult)
{
Console.WriteLine("Enter digits one after another");
for (int i = 0; i < 11; i++)
{
var p = Console.ReadLine();
await userAgent.SendDtmf(byte.Parse(p));
}
}
Console.WriteLine("Enter ?");
Console.ReadLine();
await userAgent.SendDtmf(35);
Thread.Sleep(60000);
userAgent.Hangup();
_waveFile.Dispose();
Console.WriteLine("Hangup");
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Convert Audio");
AddConsoleLogger();
//WaveFormatConversionStream converter = new WaveFormatConversionStream(_format_s16le48k, )
_waveFile = new WaveFileWriter("output_s16le48k.mp3", _format_s16le48k);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) =>
{
Console.WriteLine("Call hungup.");
_waveFile?.Close();
};
//EnableTraceLogs(sipTransport);
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioFormats = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
};
var rtpSession = new RtpAudioSession(audioOptions, audioFormats);
rtpSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"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();
SIPSorcery.Net.DNSManager.Stop();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Play Sounds Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new WindowsAudioRtpSession();
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
if (callResult)
{
await Task.Delay(1000);
await rtpSession.SendAudioFromStream(new FileStream(WELCOME_16K, FileMode.Open), AudioSamplingRatesEnum.SampleRate16KHz);
await Task.Delay(1000);
await rtpSession.SendAudioFromStream(new FileStream(GOODBYE_16K, FileMode.Open), AudioSamplingRatesEnum.SampleRate16KHz);
}
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Give the hangup a chance to complete.
await Task.Delay(1000);
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
static async Task Main()
{
Console.WriteLine("SIP Get Started");
var userAgent = new SIPUserAgent();
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("Press any key to hangup and exit.");
Console.ReadLine();
}
protected override void OnCreate(Bundle savedInstanceState)
{
base.OnCreate(savedInstanceState);
Xamarin.Essentials.Platform.Init(this, savedInstanceState);
SetContentView(Resource.Layout.activity_main);
var callButton = FindViewById <Button>(Resource.Id.callButton);
var cancelButton = FindViewById <Button>(Resource.Id.cancelButton);
var destination = FindViewById <TextView>(Resource.Id.callDestination);
var statusScroll = FindViewById <ScrollView>(Resource.Id.statusScroll);
var statusText = FindViewById <TextView>(Resource.Id.statusTextView);
Action <string> logDelegate = (str) =>
{
this.RunOnUiThread(() =>
{
statusText.Append(str);
statusScroll.FullScroll(FocusSearchDirection.Down);
});
};
SIPSorcery.LogFactory.Set(new TextViewLoggerFactory(logDelegate));
var userAgent = new SIPUserAgent();
callButton.Click += async(sender, e) =>
{
callButton.Enabled = false;
cancelButton.Enabled = true;
logDelegate($"Calling {destination.Text}...\n");
var callResult = await userAgent.Call(destination.Text, null, null, new AudioSendOnlyMediaSession());
logDelegate($"Call result {callResult}...\n");
};
cancelButton.Click += (sender, e) =>
{
cancelButton.Enabled = false;
callButton.Enabled = true;
logDelegate("Cancelled.\n");
};
}
public async Task PlaceCallMismatchedCapabilitiesUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport serverTransport = new SIPTransport();
SIPUDPChannel udpChannel = new SIPUDPChannel(IPAddress.Loopback, 0);
serverTransport.AddSIPChannel(udpChannel);
// Set up two user agents: one to answer the test call and one to place it.
SIPUserAgent userAgentServer = new SIPUserAgent(serverTransport, null);
SIPUserAgent userAgentClient = new SIPUserAgent(new SIPTransport(), null);
serverTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
logger.LogDebug("Request received: " + req.StatusLine);
var uas = userAgentServer.AcceptCall(req);
var serverAudioSession = CreateMockVoIPMediaEndPoint(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
var answerResult = await userAgentServer.Answer(uas, serverAudioSession);
logger.LogDebug($"Server agent answer result {answerResult}.");
Assert.False(answerResult);
};
var dstUri = udpChannel.GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"Attempting call to {dstUri.ToString()}.");
var clientMediaEndPoint = CreateMockVoIPMediaEndPoint(new List <AudioCodecsEnum> {
AudioCodecsEnum.G722
});
var callResult = await userAgentClient.Call(dstUri.ToString(), null, null, clientMediaEndPoint);
logger.LogDebug($"Client agent answer result {callResult }.");
Assert.False(callResult);
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new RTPMediaSession((int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
// Connect audio devices to RTP session.
WaveInEvent microphone = GetAudioInputDevice();
var speaker = GetAudioOutputDevice();
ConnectAudioDevicesToRtp(rtpSession, microphone, speaker);
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
if (callResult)
{
Console.WriteLine("Call attempt successful.");
microphone.StartRecording();
}
else
{
Console.WriteLine("Call attempt failed.");
}
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
microphone.StopRecording();
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.Microphone,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"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();
SIPSorcery.Net.DNSManager.Stop();
}
/// <summary>
/// Process user key presses.
/// </summary>
/// <param name="exit">The cancellation token to set if the user requests to quit the application.</param>
private static async Task OnKeyPress(CancellationToken exit)
{
try
{
while (!exit.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
// Place an outgoing call.
var ua = new SIPUserAgent(_sipTransport, null);
ua.ClientCallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallFailed += (uac, err) => Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
ua.ClientCallAnswered += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
ua.OnCallHungup += OnHangup;
var rtpSession = CreateRtpSession(ua, null);
var callResult = await ua.Call(DEFAULT_CALL_DESTINATION, null, null, rtpSession);
if (callResult)
{
await rtpSession.Start();
_calls.TryAdd(ua.Dialogue.CallId, ua);
}
}
else if (keyProps.KeyChar == 'd')
{
if (_calls.Count == 0)
{
Log.LogWarning("There are no active calls.");
}
else
{
var newestCall = _calls.OrderByDescending(x => x.Value.Dialogue.Inserted).First();
byte randomDtmf = (byte)Crypto.GetRandomInt(0, 15);
Log.LogInformation($"Sending DTMF {randomDtmf} to {newestCall.Key}.");
await newestCall.Value.SendDtmf(randomDtmf);
}
}
else if (keyProps.KeyChar == 'h')
{
if (_calls.Count == 0)
{
Log.LogWarning("There are no active calls.");
}
else
{
var oldestCall = _calls.OrderBy(x => x.Value.Dialogue.Inserted).First();
Log.LogInformation($"Hanging up call {oldestCall.Key}.");
oldestCall.Value.OnCallHungup -= OnHangup;
oldestCall.Value.Hangup();
_calls.TryRemove(oldestCall.Key, out _);
}
}
else if (keyProps.KeyChar == 'H')
{
if (_calls.Count == 0)
{
Log.LogWarning("There are no active calls.");
}
else
{
foreach (var call in _calls)
{
Log.LogInformation($"Hanging up call {call.Key}.");
call.Value.Hangup();
}
_calls.Clear();
}
}
else if (keyProps.KeyChar == 'l')
{
if (_calls.Count == 0)
{
Log.LogInformation("There are no active calls.");
}
else
{
Log.LogInformation("Current call list:");
foreach (var call in _calls)
{
Log.LogInformation($"{call.Key}: {call.Value.Dialogue.RemoteTarget}");
}
}
}
else if (keyProps.KeyChar == 'r')
{
if (_registrations.Count == 0)
{
Log.LogInformation("There are no active registrations.");
}
else
{
Log.LogInformation("Current registration list:");
foreach (var registration in _registrations)
{
Log.LogInformation($"{registration.Key}: is registered {registration.Value.IsRegistered}, last attempt at {registration.Value.LastRegisterAttemptAt}");
}
}
}
else if (keyProps.KeyChar == 't')
{
if (_calls.Count == 0)
{
Log.LogWarning("There are no active calls.");
}
else
{
var newestCall = _calls.OrderByDescending(x => x.Value.Dialogue.Inserted).First();
Log.LogInformation($"Transferring call {newestCall.Key} to {DEFAULT_TRANSFER_DESTINATION}.");
bool transferResult = await newestCall.Value.BlindTransfer(SIPURI.ParseSIPURI(DEFAULT_TRANSFER_DESTINATION), TimeSpan.FromSeconds(3), exit);
if (transferResult)
{
Log.LogInformation($"Transferring succeeded.");
// The remote party will often put us on hold after the transfer.
await Task.Delay(1000);
newestCall.Value.OnCallHungup -= OnHangup;
newestCall.Value.Hangup();
_calls.TryRemove(newestCall.Key, out _);
}
else
{
Log.LogWarning($"Transfer attempt failed.");
}
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
Log.LogInformation("Quitting");
break;
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception OnKeyPress. {excp.Message}.");
}
}
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 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();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Video Call Demo");
Console.WriteLine("Press ctrl-c to exit.");
Log = AddConsoleLogger();
ManualResetEvent exitMRE = new ManualResetEvent(false);
_sipTransport = new SIPTransport();
EnableTraceLogs(_sipTransport);
// Open a window to display the video feed from the remote SIP party.
_form = new Form();
_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);
Application.EnableVisualStyles();
ThreadPool.QueueUserWorkItem(delegate { Application.Run(_form); });
ManualResetEvent formMre = new ManualResetEvent(false);
_form.Activated += (object sender, EventArgs e) => formMre.Set();
Console.WriteLine("Waiting for form activation.");
formMre.WaitOne();
_sipTransport.SIPTransportRequestReceived += OnSIPTransportRequestReceived;
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.OnCallHungup += (dialog) => exitMRE.Set();
var windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudioEndPoint.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var windowsVideoEndPoint = new WindowsVideoEndPoint(new VpxVideoEncoder());
// Fallback to a test pattern source if accessing the Windows webcam fails.
var testPattern = new VideoTestPatternSource(new VpxVideoEncoder());
MediaEndPoints mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
var voipMediaSession = new VoIPMediaSession(mediaEndPoints, testPattern);
voipMediaSession.AcceptRtpFromAny = true;
windowsVideoEndPoint.OnVideoSourceRawSample += (uint durationMilliseconds, int width, int height, byte[] sample, VideoPixelFormatsEnum pixelFormat) =>
{
_form?.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = sample)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap(width, height, width * 3, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_localVideoPicBox.Image = bmpImage;
}
}
}));
};
Console.WriteLine("Starting local video source...");
await windowsVideoEndPoint.StartVideo().ConfigureAwait(false);
// Place the call and wait for the result.
Task <bool> callTask = userAgent.Call(DESTINATION, null, null, voipMediaSession);
callTask.Wait(CALL_TIMEOUT_SECONDS * 1000);
if (callTask.Result)
{
Log.LogInformation("Call attempt successful.");
windowsVideoEndPoint.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
_form?.BeginInvoke(new Action(() =>
{
unsafe
{
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;
}
}
}));
};
windowsAudioEndPoint.PauseAudio().Wait();
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Music);
}
else
{
Log.LogWarning("Call attempt failed.");
Console.WriteLine("Press ctrl-c to exit.");
}
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
exitMRE.Set();
};
exitMRE.WaitOne();
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up.");
userAgent.Hangup();
Task.Delay(1000).Wait();
}
// Clean up.
_form.BeginInvoke(new Action(() => _form.Close()));
_sipTransport.Shutdown();
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery call hold 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.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
// Check whether an override desination has been entered on the command line.
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
else
{
callUri = SIPURI.ParseSIPURIRelaxed(args[0]);
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
EnableTraceLogs(_sipTransport);
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
_ourRtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out _ourRtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
userAgent.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
exitCts.Cancel();
};
userAgent.ClientCallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
// Only set the remote RTP end point if there hasn't already been a packet received on it.
if (_remoteRtpEndPoint == null)
{
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
userAgent.CallHungup += () =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.OnReinviteRequest += ReinviteRequestReceived;
// The only incoming requests that need to be explicitly in this example program are in-dialog
// re-INVITE requests that are being used to place the call on/off hold.
_sipTransport.SIPTransportRequestReceived += (localSIPEndPoint, remoteEndPoint, sipRequest) =>
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
_sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception excp)
{
Log.LogError($"Exception processing request. {excp.Message}");
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => RecvRtp(_ourRtpSocket, rtpRecvSession, exitCts));
Task.Run(() => SendRtp(_ourRtpSocket, rtpSendSession, exitCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri.ToString(),
$"sip:{SIP_USERNAME}@localhost",
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
_ourSDP.ToString(),
null);
userAgent.Call(callDescriptor);
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsAnswered)
{
if (_holdStatus == HoldStatus.None)
{
Log.LogInformation("Placing the remote call party on hold.");
_holdStatus = HoldStatus.WePutOnHold;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDONLY);
userAgent.SendReInviteRequest(_ourSDP);
}
else if (_holdStatus == HoldStatus.WePutOnHold)
{
Log.LogInformation("Removing the remote call party from hold.");
_holdStatus = HoldStatus.None;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
userAgent.SendReInviteRequest(_ourSDP);
}
else
{
Log.LogInformation("Sorry we're already on hold by the remote call party.");
}
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"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...");
_ourRtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && userAgent != null)
{
if (userAgent.IsAnswered)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {userAgent?.CallDescriptor?.To}.");
userAgent.Cancel();
}
// 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 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($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, err, resp) =>
{
Console.WriteLine($"Call failed {err}");
_waveFile?.Close();
};
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
else
{
Console.WriteLine("Cancelling call");
userAgent.Cancel();
}
};
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
/// <summary>
/// Process user key presses.
/// </summary>
/// <param name="exit">The cancellation token to set if the user requests to quit the application.</param>
private static void OnKeyPress(CancellationTokenSource exitCts)
{
try
{
while (!exitCts.IsCancellationRequested)
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
SIPUserAgent ua = null;
string dst = null;
if (_transfereeCall == null)
{
dst = TRANSFEREE_DST;
_transfereeCall = new SIPUserAgent(_sipTransport, null);
ua = _transfereeCall;
}
else if (_targetCall == null)
{
dst = TARGET_DST;
_targetCall = new SIPUserAgent(_sipTransport, null);
ua = _targetCall;
}
if (ua == null)
{
Log.LogWarning("Cannot place a new call, both the transferee and target user agents are busy.");
}
else
{
// Place an outgoing call.
ua.ClientCallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallFailed += (uac, err, resp) => Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}, Status code: {resp?.StatusCode}");
ua.ClientCallAnswered += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.OnDtmfTone += (key, duration) => Log.LogInformation($"Received DTMF tone {key}.");
//ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"transferee rtp event {evt.EventID}, ssrc {hdr.SyncSource}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
ua.OnCallHungup += (dialog) => Log.LogDebug("Call hungup by remote party.");
Task.Run(async() =>
{
var rtpSession = CreateRtpSession();
var callResult = await ua.Call(dst, null, null, rtpSession);
if (!callResult)
{
Log.LogWarning($"Call to {dst} failed.");
}
else
{
Log.LogInformation($"Call to {dst} was successful.");
}
});
}
}
else if (keyProps.KeyChar == 'h')
{
if (_transfereeCall != null)
{
Log.LogDebug("Hanging up transferee call.");
_transfereeCall.Hangup();
}
if (_targetCall != null)
{
Log.LogDebug("Hanging up target call.");
_targetCall.Hangup();
}
_transfereeCall = null;
_targetCall = null;
}
else if (keyProps.KeyChar == 't')
{
if (_transfereeCall == null)
{
Log.LogWarning("The call to the transferee is not established.");
}
else if (_targetCall == null)
{
Log.LogWarning("The call to the target is not established.");
}
else
{
Task.Run(async() =>
{
Log.LogInformation("Initiating transfer to the transferee...");
bool transferResult = await _transfereeCall.AttendedTransfer(_targetCall.Dialogue, TimeSpan.FromSeconds(2), exitCts.Token);
Log.LogDebug($"Transfer result {transferResult}.");
await Task.Delay(2000);
Log.LogDebug($"Transferee call status {_transfereeCall?.IsCallActive}.");
Log.LogDebug($"Target call status {_targetCall?.IsCallActive}.");
});
}
}
else if (keyProps.KeyChar == 'a')
{
Log.LogDebug($"Yes I am alive!");
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
Log.LogInformation("Quitting");
exitCts.Cancel();
break;
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception OnKeyPress. {excp.Message}.");
}
}
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();
}
}
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();
}
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 async Task Main()
{
Console.WriteLine("SIPSorcery Send DTMF Tones example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, rtpSession);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
if (callResult)
{
Console.WriteLine("Call attempt successful.");
// Give the call some time to answer.
await Task.Delay(1000);
// Send the DTMF tones.
await userAgent.SendDtmf(0x05);
await Task.Delay(2000);
await userAgent.SendDtmf(0x09);
await Task.Delay(2000);
await userAgent.SendDtmf(0x02);
await Task.Delay(2000);
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
rtpCts.Cancel();
userAgent.Hangup();
}
}
else
{
Console.WriteLine("Call attempt failed.");
}
Log.LogInformation("Exiting...");
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Play Sounds Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, OUTBOUND_PROXY);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
userAgent.ClientCallFailed += (uac, error, sipResponse) => exitCts.Cancel();
userAgent.OnCallHungup += (dialog) => exitCts.Cancel();
var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
//windowsAudio.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
//voipMediaSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 20;
//voipMediaSession.AudioLocalTrack.Capabilities.Clear();
//voipMediaSession.AudioLocalTrack.Capabilities.Add(
// new SDPAudioVideoMediaFormat(new AudioFormat(AudioCodecsEnum.L16, 118, 8000)));
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();
};
// Place the call and wait for the result.
var callTask = userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
Console.WriteLine($"Call to {DESTINATION} succeeded.");
await windowsAudio.PauseAudio();
try
{
await voipMediaSession.AudioExtrasSource.StartAudio();
//Console.WriteLine("Sending welcome message from 8KHz sample.");
await voipMediaSession.AudioExtrasSource.SendAudioFromStream(new FileStream(WELCOME_8K, FileMode.Open), AudioSamplingRatesEnum.Rate8KHz);
await Task.Delay(200, exitCts.Token);
Console.WriteLine("Sending sine wave.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.SineWave);
await Task.Delay(5000, exitCts.Token);
Console.WriteLine("Sending white noise signal.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.WhiteNoise);
await Task.Delay(2000, exitCts.Token);
Console.WriteLine("Sending pink noise signal.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.PinkNoise);
await Task.Delay(2000, exitCts.Token);
Console.WriteLine("Sending silence.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Silence);
await Task.Delay(2000, exitCts.Token);
Console.WriteLine("Playing music.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Music);
await Task.Delay(5000, exitCts.Token);
Console.WriteLine("Sending goodbye message from 16KHz sample.");
await voipMediaSession.AudioExtrasSource.SendAudioFromStream(new FileStream(GOODBYE_16K, FileMode.Open), AudioSamplingRatesEnum.Rate16KHz);
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.None);
await voipMediaSession.AudioExtrasSource.PauseAudio();
await Task.Delay(200, exitCts.Token);
}
catch (System.Threading.Tasks.TaskCanceledException)
{ }
// Switch to the external microphone input source.
await windowsAudio.ResumeAudio();
exitCts.Token.WaitHandle.WaitOne();
}
else
{
Console.WriteLine($"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();
}