/// <summary>
/// Places an outgoing SIP call.
/// </summary>
/// <param name="destination">The SIP URI to place a call to. The destination can be a full SIP URI in which case the all will
/// be placed anonymously directly to that URI. Alternatively it can be just the user portion of a URI in which case it will
/// be sent to the configured SIP server.</param>
public async Task Call(string destination)
{
// Determine if this is a direct anonymous call or whether it should be placed using the pre-configured SIP server account.
SIPURI callURI = null;
string sipUsername = null;
string sipPassword = null;
string fromHeader = null;
if (destination.Contains("@") || m_sipServer == null)
{
// Anonymous call direct to SIP server specified in the URI.
callURI = SIPURI.ParseSIPURIRelaxed(destination);
fromHeader = (new SIPFromHeader(m_sipFromName, SIPURI.ParseSIPURI(SIPFromHeader.DEFAULT_FROM_URI), null)).ToString();
}
else
{
// This call will use the pre-configured SIP account.
callURI = SIPURI.ParseSIPURIRelaxed(destination + "@" + m_sipServer);
sipUsername = m_sipUsername;
sipPassword = m_sipPassword;
fromHeader = (new SIPFromHeader(m_sipFromName, new SIPURI(m_sipUsername, m_sipServer, null), null)).ToString();
}
StatusMessage(this, $"Starting call to {callURI}.");
var lookupResult = await Task.Run(() =>
{
return(SIPDNSManager.ResolveSIPService(callURI, false));
});
if (lookupResult == null || lookupResult.LookupError != null)
{
StatusMessage(this, $"Call failed, could not resolve {callURI}.");
}
else
{
var dstEndpoint = lookupResult.GetSIPEndPoint();
StatusMessage(this, $"Call progressing, resolved {callURI} to {dstEndpoint}.");
System.Diagnostics.Debug.WriteLine($"DNS lookup result for {callURI}: {dstEndpoint}.");
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, null, null);
m_rtpMediaSessionManager.Create(dstEndpoint.Address.AddressFamily);
m_rtpMediaSessionManager.RTPMediaSession.RemotePutOnHold += OnRemotePutOnHold;
m_rtpMediaSessionManager.RTPMediaSession.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.InitiateCall(callDescriptor, m_rtpMediaSessionManager.RTPMediaSession);
}
}
static void Main()
{
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();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
// Get the default speaker.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
WaveInEvent waveInEvent = GetAudioInputDevice();
RTPMediaSession RtpMediaSession = 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.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}.");
PlayRemoteMedia(RtpMediaSession, audioOutProvider);
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
hasCallFailed = true;
exitCts.Cancel();
}
};
userAgent.OnCallHungup += () =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller.");
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);
RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
await userAgent.Answer(incomingCall, RtpMediaSession);
PlayRemoteMedia(RtpMediaSession, audioOutProvider);
waveInEvent.StartRecording();
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);
}
};
// Wire up the RTP send session to the audio output device.
uint rtpSendTimestamp = 0;
waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) =>
{
byte[] sample = new byte[args.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < args.BytesRecorded; index += 2)
{
var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
if (RtpMediaSession != null)
{
RtpMediaSession.SendAudioFrame(rtpSendTimestamp, sample);
rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds);
}
};
// 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)
{
RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
var callDescriptor = GetCallDescriptor(DEFAULT_DESTINATION_SIP_URI);
await userAgent.InitiateCall(callDescriptor, RtpMediaSession);
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (RtpMediaSession.LocalOnHold)
{
Log.LogInformation("Taking the remote call party off hold.");
RtpMediaSession.TakeOffHold();
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
RtpMediaSession.PutOnHold();
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 't')
{
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...");
RtpMediaSession?.Close();
waveInEvent?.StopRecording();
audioOutEvent?.Stop();
if (!isCallHungup && userAgent != null)
{
if (userAgent.IsCallActive)
{
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
}