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");
}
/// <summary>
/// Requests the RTP session to send a RTP event representing a DTMF tone to the
/// remote party.
/// </summary>
/// <param name="tone">A byte representing the tone to send. Must be between 0 and 15.</param>
public Task SendDTMF(byte tone)
{
if (m_userAgent != null)
{
return(m_userAgent.SendDtmf(tone));
}
else
{
return(Task.FromResult(0));
}
}
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();
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
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)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
ua.OnCallHungup += OnHangup;
ua.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
ua.OnTransactionTraceMessage += (tx, msg) => Log.LogDebug($"uas tx {tx.TransactionId}: {msg}");
ua.ServerCallRingTimeout += (uas) =>
{
Log.LogWarning($"Incoming call timed out in {uas.ClientTransaction.TransactionState} state waiting for client ACK, terminating.");
ua.Hangup();
};
var uas = ua.AcceptCall(sipRequest);
var rtpSession = CreateRtpSession(ua);
// Insert a brief delay to allow testing of the "Ringing" progress response.
// Without the delay the call gets answered before it can be sent.
await Task.Delay(500);
await ua.Answer(uas, rtpSession);
if (ua.IsCallActive)
{
_calls.TryAdd(ua.Dialogue.CallId, ua);
if (sipRequest.URI.User != null)
{
if (Int32.TryParse(sipRequest.URI.User, out int dtmfCode))
{
Log.LogDebug($"URI dtmf code {dtmfCode}.");
while (dtmfCode > 0)
{
byte dtmfByte = (byte)(dtmfCode % 10);
Log.LogDebug($"Sending DTMF {dtmfByte} to caller.");
if (!ua.IsCallActive)
{
Log.LogWarning($"Client call no longer active.");
break;
}
else
{
await ua.SendDtmf(dtmfByte);
}
dtmfCode /= 10;
}
}
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
static void Main()
{
Console.WriteLine("SIPSorcery Blind Transfer Demo: Transferee");
Console.WriteLine("Waiting for incoming call from Transferor.");
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.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(_sipTransport);
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => Log.LogDebug("Call hungup by remote party.");
userAgent.OnIncomingCall += async(ua, req) =>
{
var rtpAudioSession = new AudioSendOnlyMediaSession();
var uas = ua.AcceptCall(req);
bool answerResult = await ua.Answer(uas, rtpAudioSession);
Log.LogDebug($"Answer incoming call result {answerResult}.");
_ = Task.Run(async() =>
{
await Task.Delay(1000);
Log.LogDebug($"Sending DTMF sequence {string.Join("", DTMF_SEQUENCEFOR_TRANSFEROR.Select(x => x))}.");
foreach (byte dtmf in DTMF_SEQUENCEFOR_TRANSFEROR)
{
Log.LogDebug($"Sending DTMF tone to transferor {dtmf}.");
await ua.SendDtmf(dtmf);
}
});
};
userAgent.OnTransferRequested += (referredTo, referredBy) => true;
userAgent.OnTransferToTargetSuccessful += (dst) =>
{
Task.Run(async() =>
{
await Task.Delay(1000);
Log.LogDebug($"Sending DTMF sequence {string.Join("", DTMF_SEQUENCEFOR_TRANSFEROR.Select(x => x))}.");
foreach (byte dtmf in DTMF_SEQUENCEFOR_TRANSFEROR)
{
Log.LogDebug($"Sending DTMF tone to target {dtmf}.");
await userAgent.SendDtmf(dtmf);
}
});
};
Task.Run(() => OnKeyPress(userAgent, exitCts));
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
//userAgent?.Hangup();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
if (_sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
_sipTransport.Shutdown();
}
#endregion
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, OUTBOUND_PROXY);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
userAgent.OnCallHungup += (dialog) => exitCts.Cancel();
var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;
string fromHeader = (new SIPFromHeader(USERNAME, new SIPURI(USERNAME, DOMAIN, null), null)).ToString();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(USERNAME, PASSWORD, DESTINATION, fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, null, null);
//callDescriptor.CallId = "16152412565";
//callDescriptor.AuthUsername = USERNAME;
// Place the call and wait for the result.
var callTask = userAgent.Call(callDescriptor, voipMediaSession);
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();
_waveFile.Dispose();
}
}
;
exitCts.Cancel();
};
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
Console.WriteLine("Enter digits one after another");
string meetingNo = "1711622132";
Console.WriteLine("Enter meetingno ?");
Console.ReadLine();
foreach (var item in meetingNo)
{
await userAgent.SendDtmf(byte.Parse(item.ToString()));
Console.WriteLine("Sending DTMF - " + byte.Parse(item.ToString()));
Thread.Sleep(2000);
}
Thread.Sleep(2000);
await userAgent.SendDtmf(Encoding.ASCII.GetBytes("#")[0]);
Thread.Sleep(13000);
Console.WriteLine("Sending AttendeeID ?");
/*string attendeeId = "635619";
* foreach (var item in attendeeId)
* {
* await userAgent.SendDtmf(byte.Parse(item.ToString()));
* Console.WriteLine("Sending DTMF - " + byte.Parse(item.ToString()));
* Thread.Sleep(2000);
* }*/
await userAgent.SendDtmf(Encoding.ASCII.GetBytes("#")[0]);
Console.ReadLine();
await userAgent.SendDtmf(Encoding.ASCII.GetBytes("#")[0]);
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();
}
