/// <summary>
/// Event handler for RTP events from the remote call party. Fires
/// when the event is completed.
/// </summary>
/// <param name="rtpEvent">The received RTP event.</param>
private void OnRemoteRtpEvent(RTPEvent rtpEvent)
{
if (rtpEvent.EndOfEvent)
{
OnDtmfTone?.Invoke(rtpEvent.EventID, rtpEvent.Duration);
}
}
/// <summary>
/// Event handler for RTP events from the remote call party. Fires
/// when the event is completed.
/// </summary>
/// <param name="rtpEvent">The received RTP event.</param>
/// <param name="rtpHeader">THe RTP header on the packet that the event was received in.</param>
private void OnRemoteRtpEvent(RTPEvent rtpEvent, RTPHeader rtpHeader)
{
OnRtpEvent?.Invoke(rtpEvent, rtpHeader);
if (rtpEvent.EndOfEvent)
{
OnDtmfTone?.Invoke(rtpEvent.EventID, rtpEvent.Duration);
}
}
/// <summary>
/// Event handler for RTP events from the remote call party. Fires
/// when the event is completed.
/// </summary>
/// <param name="rtpEvent">The received RTP event.</param>
private void OnRemoteRtpEvent(RTPEvent rtpEvent)
{
if (lastRtpEventEndTimeStamp != rtpEvent.Source.Header.Timestamp &&
rtpEvent.EndOfEvent)
{
lastRtpEventEndTimeStamp = rtpEvent.Source.Header.Timestamp;
OnDtmfTone?.Invoke(rtpEvent.EventID, rtpEvent.Duration);
}
}
/// <summary>
/// Event handler for RTP events from the remote call party.
/// </summary>
/// <param name="rtpEvent">The received RTP event.</param>
private void OnRemoteRtpEvent(RTPEvent rtpEvent)
{
if (rtpEvent.EndOfEvent)
{
remoteDtmfDuration = 0;
}
else if (remoteDtmfDuration == 0)
{
remoteDtmfDuration = rtpEvent.Duration;
DtmfCompleted?.Invoke(rtpEvent.EventID);
}
}
public Task SendDtmf(byte key, CancellationToken cancellationToken = default)
{
var dtmfEvent = new RTPEvent(key, false, RTPEvent.DEFAULT_VOLUME, DTMF_EVENT_DURATION, DTMF_EVENT_PAYLOAD_ID);
return(SendDtmfEvent(dtmfEvent, cancellationToken));
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent 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.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
// Un/comment this line to see/hide each SIP message sent and received.
EnableTraceLogs(sipTransport);
// Note this relies on the callURI host being an IP address. If it's a hostname a DNS lookup is required.
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(callUri.ToSIPEndPoint().Address);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var rtpSession = new RTPSession((int)SDPMediaFormatsEnum.PCMU, null, null, true, localIPAddress.AddressFamily);
var offerSDP = rtpSession.GetSDP(localIPAddress);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
rtpSession.DestinationEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {rtpSession.DestinationEndPoint}.");
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(okResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
}
};
// Wire up the RTP receive session to the default speaker.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
rtpSession.OnReceivedSampleReady += (sample) =>
{
for (int index = 0; index < sample.Length; index++)
{
short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]);
byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
audioOutProvider.AddSamples(pcmSample, 0, 2);
}
};
// Send audio packets (in this case silence) to the callee.
Task.Run(() => SendSilence(rtpSession, rtpCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
offerSDP.ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
// Give the call some time to answer.
Task.Delay(3000).Wait();
// Send some DTMF key presses via RTP events.
var dtmf5 = new RTPEvent(0x05, false, RTPEvent.DEFAULT_VOLUME, 1200, RTPSession.DTMF_EVENT_PAYLOAD_ID);
rtpSession.SendDtmfEvent(dtmf5, rtpCts.Token).Wait();
Task.Delay(2000, rtpCts.Token).Wait();
var dtmf9 = new RTPEvent(0x09, false, RTPEvent.DEFAULT_VOLUME, 1200, RTPSession.DTMF_EVENT_PAYLOAD_ID);
rtpSession.SendDtmfEvent(dtmf9, rtpCts.Token).Wait();
Task.Delay(2000, rtpCts.Token).Wait();
var dtmf2 = new RTPEvent(0x02, false, RTPEvent.DEFAULT_VOLUME, 1200, RTPSession.DTMF_EVENT_PAYLOAD_ID);
rtpSession.SendDtmfEvent(dtmf2, rtpCts.Token).Wait();
Task.Delay(2000, rtpCts.Token).ContinueWith((task) => { }).Wait(); // Don't care about the exception if the cancellation token is set.
Log.LogInformation("Exiting...");
rtpCts.Cancel();
audioOutEvent?.Stop();
rtpSession.CloseSession(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.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();
}
}
