/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static AudioSendOnlyMediaSession CreateRtpSession()
{
var rtpAudioSession = new AudioSendOnlyMediaSession();
// Wire up the event handler for RTP packets received from the remote party.
//rtpAudioSession.OnRtpPacketReceived += OnRtpPacketReceived;
rtpAudioSession.OnRtcpBye += (reason) => Log.LogDebug($"RTCP BYE received.");
rtpAudioSession.OnRtpClosed += (reason) => Log.LogDebug("RTP session closed.");
//rtpAudioSession.OnReceiveReport += RtpSession_OnReceiveReport;
//rtpAudioSession.OnSendReport += RtpSession_OnSendReport;
//rtpAudioSession.OnTimeout += (mediaType) =>
//{
// if (ua?.Dialogue != null)
// {
// Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
// }
// else
// {
// Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
// }
// ua.Hangup();
//};
return(rtpAudioSession);
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static AudioSendOnlyMediaSession CreateRtpSession(SIPUserAgent ua)
{
var rtpAudioSession = new AudioSendOnlyMediaSession();
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}
static void Main()
{
Console.WriteLine("SIPSorcery On Hold Demo: Callee");
Console.WriteLine("Waiting for incoming call from Caller.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(_sipTransport);
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => Log.LogDebug("Call hungup by remote party.");
userAgent.RemotePutOnHold += () => Log.LogDebug("Remote party placed call on hold.");
userAgent.RemoteTookOffHold += () => Log.LogDebug("Remote party took call off hold.");
userAgent.OnIncomingCall += async(ua, req) =>
{
var rtpAudioSession = new AudioSendOnlyMediaSession(null, _rtpPort);
_rtpPort += 2;
rtpAudioSession.OnReceiveReport += RtpSession_OnReceiveReport;
//rtpAudioSession.OnSendReport += RtpSession_OnSendReport;
var uas = ua.AcceptCall(req);
bool answerResult = await ua.Answer(uas, rtpAudioSession);
Log.LogDebug($"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);
}
});
};
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
}
