/// <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 RtpAudioSession CreateRtpSession()
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
// Wire up the event handler for RTP packets received from the remote party.
//rtpAudioSession.OnRtpPacketReceived += (type, rtp) => OnRtpPacketReceived(type, rtp);
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($"RTP timeout on incomplete call, closing RTP session.");
// }
// ua.Hangup();
//};
return(rtpAudioSession);
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua)
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioSource = AudioSourcesEnum.SineWave;
if (string.IsNullOrEmpty(dst) || !Enum.TryParse <AudioSourcesEnum>(dst, out audioSource))
{
audioSource = AudioSourcesEnum.Silence;
}
var audioOptions = new AudioSourceOptions {
AudioSource = audioSource
};
if (audioSource == AudioSourcesEnum.Music)
{
audioOptions.SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>();
if (codecs.Contains(SDPMediaFormatsEnum.PCMA))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.PCMA, MUSIC_FILE_PCMA);
}
if (codecs.Contains(SDPMediaFormatsEnum.PCMU))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.PCMU, MUSIC_FILE_PCMU);
}
if (codecs.Contains(SDPMediaFormatsEnum.G722))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.G722, MUSIC_FILE_G722);
}
}
;
Log.LogInformation($"RTP audio session source set to {audioSource}.");
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (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);
}
/// <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();
}
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($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
static void Main()
{
Console.WriteLine("SIPSorcery Blind Transfer Demo: Transferee");
Console.WriteLine("Waiting for incoming call from Transferor.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(_sipTransport);
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => Log.LogDebug("Call hungup by remote party.");
userAgent.OnIncomingCall += async(ua, req) =>
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
var uas = ua.AcceptCall(req);
bool answerResult = await ua.Answer(uas, rtpAudioSession);
Log.LogDebug($"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();
SIPSorcery.Net.DNSManager.Stop();
if (_sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
_sipTransport.Shutdown();
}
#endregion
}
/// <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, Scenarios scenario)
{
//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.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
var result = await ua.Call(dst.ToString(), null, null, voipMediaSession);
if (scenario == Scenarios.uacw)
{
// Wait for the remote party to hangup the call.
logger.LogDebug("Waiting for the remote party to hangup the call...");
ManualResetEvent hangupMre = new ManualResetEvent(false);
ua.OnCallHungup += (dialog) => hangupMre.Set();
hangupMre.WaitOne();
logger.LogDebug("Call hungup by remote party.");
}
else
{
// We hangup the call after 1s.
await Task.Delay(1000);
ua.Hangup();
}
// Need a bit of time for the BYE to complete.
await Task.Delay(500);
return(result);
}
catch (Exception excp)
{
logger.LogError($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
