static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started PortAudio Demo (YMMV)");
AddConsoleLogger();
var userAgent = new SIPUserAgent();
var portAudioEndPoint = new PortAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(portAudioEndPoint.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
await Task.Delay(1000);
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSink = winAudio, AudioSource = winAudio
});
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"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 void Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
Log = AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
userAgent.OnIncomingCall += async(ua, req) =>
{
WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession voipMediaSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;
var uas = userAgent.AcceptCall(req);
await userAgent.Answer(uas, voipMediaSession);
};
Console.WriteLine("press any key to exit...");
Console.Read();
// Clean up.
_sipTransport.Shutdown();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, err, resp) =>
{
Console.WriteLine($"Call failed {err}");
_waveFile?.Close();
};
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession voipSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
voipSession.OnRtpPacketReceived += OnRtpPacketReceived;
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
else
{
Console.WriteLine("Cancelling call");
userAgent.Cancel();
}
};
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipSession);
Console.WriteLine($"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();
}
/// <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("SIP Get Started");
var userAgent = new SIPUserAgent();
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("Press any key to hangup and exit.");
Console.ReadLine();
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <AudioCodecsEnum> codecs = new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA, AudioCodecsEnum.G722
};
var audioSource = AudioSourcesEnum.SineWave;
if (string.IsNullOrEmpty(dst) || !Enum.IsDefined(typeof(AudioSourcesEnum), dst) || !Enum.TryParse(dst, out audioSource))
{
audioSource = AudioSourcesEnum.Music;
}
Log.LogInformation($"RTP audio session source set to {audioSource}.");
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = audioSource
});
audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec));
var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpAudioSession.AcceptRtpFromAny = true;
// 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>
/// Creates the media session to use with the SIP call.
/// </summary>
/// <param name="audioSrcOpts">The audio source options to set when the call is first
/// answered. These options can be adjusted afterwards to do things like put play
/// on hold music etc.</param>
/// <returns>A new media session object.</returns>
private VoIPMediaSession CreateMediaSession()
{
var windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder(), m_audioOutDeviceIndex);
var windowsVideoEndPoint = new WindowsVideoEndPoint();
MediaEndPoints mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
return(voipMediaSession);
}
/// <summary>
/// Creates the media session to use with the SIP call.
/// </summary>
/// <param name="audioSrcOpts">The audio source options to set when the call is first
/// answered. These options can be adjusted afterwards to do things like put play
/// on hold music etc.</param>
/// <returns>A new media session object.</returns>
private VoIPMediaSession CreateMediaSession()
{
var windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder(), m_audioOutDeviceIndex);
var windowsVideoEndPoint = new WindowsVideoEndPoint();
MediaEndPoints mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
// Fallback video source if a Windows webcam cannot be accessed.
var testPatternSource = new VideoTestPatternSource(new VideoEncoder());
var voipMediaSession = new VoIPMediaSession(mediaEndPoints, testPatternSource);
// voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;
return(voipMediaSession);
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
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.
Log = AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
winAudio.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogInformation($"Incoming call from {req.Header.From.FriendlyDescription()} at {req.RemoteSIPEndPoint}.");
var uas = userAgent.AcceptCall(req);
if (userAgent?.IsCallActive == true)
{
// If we are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request.");
uas.Reject(SIPResponseStatusCodesEnum.BusyHere, null);
}
else
{
var voipSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipSession.AcceptRtpFromAny = true;
var answerResult = await userAgent.Answer(uas, voipSession);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
var voipSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipSession.AcceptRtpFromAny = true;
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, voipSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
(userAgent.MediaSession as VoIPMediaSession).TakeOffHold();
userAgent.TakeOffHold();
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
await(userAgent.MediaSession as VoIPMediaSession).PutOnHold();
userAgent.PutOnHold();
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
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($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"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...");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
ManualResetEvent exitMre = new ManualResetEvent(false);
bool isCallHungup = false;
bool hasCallFailed = false;
Log = AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0], out callUri))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var audioSession = new WindowsAudioEndPoint(new AudioEncoder());
var rtpSession = new VoIPMediaSession(audioSession.ToMediaEndPoints());
var offerSDP = rtpSession.CreateOffer(null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += async(uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
if (resp.Status == SIPResponseStatusCodesEnum.SessionProgress)
{
await rtpSession.Start();
}
};
uac.CallFailed += (uac, err, resp) =>
{
Log.LogWarning($"Call attempt to {uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += async(iuac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body));
if (result == SetDescriptionResultEnum.OK)
{
await rtpSession.Start();
}
else
{
Log.LogWarning($"Failed to set remote description {result}.");
uac.Hangup();
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(okResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
exitMre.Set();
}
}
};
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
offerSDP.ToString(),
null);
uac.Call(callDescriptor, null);
uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
rtpSession.Close(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();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
Log = AddConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2(SIPS_CERTIFICATE_PATH);
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
VoIPMediaSession rtpSession = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Music
});
rtpSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = extrasSource
});
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
await sipTransport.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
rtpCts = new CancellationTokenSource();
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.Close(null);
};
rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
await rtpSession.Start();
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Log.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.Close(null);
rtpCts?.Cancel();
}
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}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpSession?.Close(null);
rtpCts?.Cancel();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpSession?.Close(null);
rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
Log.LogInformation("Quitting...");
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Send DTMF Tones example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
Log = AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
Console.WriteLine($"Calling {DEFAULT_DESTINATION_SIP_URI}.");
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, voipMediaSession);
// 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();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Play Sounds 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($"Call failed {error}.");
userAgent.ClientCallFailed += (uac, error, sipResponse) => exitCts.Cancel();
userAgent.OnCallHungup += (dialog) => exitCts.Cancel();
var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
//windowsAudio.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
//voipMediaSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 20;
//voipMediaSession.AudioLocalTrack.Capabilities.Clear();
//voipMediaSession.AudioLocalTrack.Capabilities.Add(
// new SDPAudioVideoMediaFormat(new AudioFormat(AudioCodecsEnum.L16, 118, 8000)));
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent != null)
{
if (userAgent.IsCalling || userAgent.IsRinging)
{
Console.WriteLine("Cancelling in progress call.");
userAgent.Cancel();
}
else if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up established call.");
userAgent.Hangup();
}
}
;
exitCts.Cancel();
};
// Place the call and wait for the result.
var callTask = userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
Console.WriteLine($"Call to {DESTINATION} succeeded.");
await windowsAudio.PauseAudio();
try
{
await voipMediaSession.AudioExtrasSource.StartAudio();
//Console.WriteLine("Sending welcome message from 8KHz sample.");
await voipMediaSession.AudioExtrasSource.SendAudioFromStream(new FileStream(WELCOME_8K, FileMode.Open), AudioSamplingRatesEnum.Rate8KHz);
await Task.Delay(200, exitCts.Token);
Console.WriteLine("Sending sine wave.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.SineWave);
await Task.Delay(5000, exitCts.Token);
Console.WriteLine("Sending white noise signal.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.WhiteNoise);
await Task.Delay(2000, exitCts.Token);
Console.WriteLine("Sending pink noise signal.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.PinkNoise);
await Task.Delay(2000, exitCts.Token);
Console.WriteLine("Sending silence.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Silence);
await Task.Delay(2000, exitCts.Token);
Console.WriteLine("Playing music.");
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Music);
await Task.Delay(5000, exitCts.Token);
Console.WriteLine("Sending goodbye message from 16KHz sample.");
await voipMediaSession.AudioExtrasSource.SendAudioFromStream(new FileStream(GOODBYE_16K, FileMode.Open), AudioSamplingRatesEnum.Rate16KHz);
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.None);
await voipMediaSession.AudioExtrasSource.PauseAudio();
await Task.Delay(200, exitCts.Token);
}
catch (System.Threading.Tasks.TaskCanceledException)
{ }
// Switch to the external microphone input source.
await windowsAudio.ResumeAudio();
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();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Video Call Demo");
Console.WriteLine("Press ctrl-c to exit.");
Log = AddConsoleLogger();
ManualResetEvent exitMRE = new ManualResetEvent(false);
var sipTransport = new SIPTransport();
sipTransport.EnableTraceLogs();
var userAgent = new SIPUserAgent(sipTransport, null, true);
#region Set up a simple Windows Form with two picture boxes.
_form = new Form();
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_localVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, 0),
Visible = true
};
_remoteVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, VIDEO_FRAME_HEIGHT),
Visible = true
};
_form.Controls.Add(_localVideoPicBox);
_form.Controls.Add(_remoteVideoPicBox);
#endregion
Application.EnableVisualStyles();
ThreadPool.QueueUserWorkItem(delegate { Application.Run(_form); });
_form.FormClosing += (sender, e) => _isFormActivated = false;
_form.Activated += (sender, e) => _isFormActivated = true;
_form.FormClosed += (sender, e) => userAgent.Hangup();
userAgent.OnCallHungup += (dialog) =>
{
if (_isFormActivated)
{
_form.Close();
}
};
// Video sink and source to generate and consume VP8 video streams.
var testPattern = new VideoTestPatternSource(new VpxVideoEncoder());
var vp8VideoSink = new VideoEncoderEndPoint();
// Add the video sink and source to the media session.
MediaEndPoints mediaEndPoints = new MediaEndPoints
{
VideoSink = vp8VideoSink,
VideoSource = testPattern,
};
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
#region Connect the video frames generate from the sink and source to the Windows form.
testPattern.OnVideoSourceRawSample += (uint durationMilliseconds, int width, int height, byte[] sample, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
unsafe
{
fixed(byte *s = sample)
{
var bmpImage = new Bitmap(width, height, width * 3, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_localVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
vp8VideoSink.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
unsafe
{
fixed(byte *s = bmp)
{
var bmpImage = new Bitmap((int)width, (int)height, stride, PixelFormat.Format24bppRgb, (IntPtr)s);
_remoteVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
#endregion
// Place the call.
var callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession).ConfigureAwait(false);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("Press any key to hangup and exit.");
Console.ReadLine();
if (userAgent.IsCallActive)
{
_isFormActivated = false;
userAgent.Hangup();
await Task.Delay(1000).ConfigureAwait(false);
}
sipTransport.Shutdown();
}
/// <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($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
static void Main()
{
Console.WriteLine("SIPSorcery Attended Transfer example.");
Console.WriteLine("Press 'c' to place a call to the default destination.");
Console.WriteLine("Place two simultaneous SIP calls to this program and then press 't'.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
Log = AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
// Create two user agents. Each gets configured to answer an incoming call.
var userAgent1 = new SIPUserAgent(sipTransport, null);
var userAgent2 = new SIPUserAgent(sipTransport, null);
userAgent1.OnCallHungup += (dialog) => Log.LogInformation($"UA1: Call hungup by remote party.");
userAgent1.ServerCallCancelled += (uas) => Log.LogInformation("UA1: Incoming call cancelled by caller.");
userAgent2.OnCallHungup += (dialog) => Log.LogInformation($"UA2: Call hungup by remote party.");
userAgent2.ServerCallCancelled += (uas) => Log.LogInformation("UA2: Incoming call cancelled by caller.");
userAgent2.OnTransferNotify += (sipFrag) =>
{
if (!string.IsNullOrEmpty(sipFrag))
{
Log.LogInformation($"UA2: Transfer status update: {sipFrag.Trim()}.");
if (sipFrag?.Contains("SIP/2.0 200") == true)
{
// The transfer attempt got a successful answer. Can hangup the call.
userAgent2.Hangup();
exitCts.Cancel();
}
}
};
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 (!userAgent1.IsCallActive || !userAgent2.IsCallActive)
{
SIPUserAgent activeAgent = (!userAgent1.IsCallActive) ? userAgent1 : userAgent2;
string agentDesc = (!userAgent1.IsCallActive) ? "UA1" : "UA2";
Log.LogInformation($"{agentDesc}: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = activeAgent.AcceptCall(sipRequest);
WindowsAudioEndPoint winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
await activeAgent.Answer(incomingCall, voipMediaSession);
Log.LogInformation($"{agentDesc}: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
else
{
// If both user agents are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
}
else
{
Log.LogDebug($"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);
}
};
// 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')
{
// Place an outgoing call using the first free user agent.
SIPUserAgent freeAgent = (!userAgent1.IsCallActive) ? userAgent1 : (!userAgent2.IsCallActive) ? userAgent2 : null;
if (freeAgent != null)
{
WindowsAudioEndPoint winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
bool callResult = await freeAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, voipMediaSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("Both user agents are already on calls.");
}
}
if (keyProps.KeyChar == 't')
{
// Initiate the attended transfer.
if (userAgent1.IsCallActive && userAgent2.IsCallActive)
{
bool result = await userAgent2.AttendedTransfer(userAgent1.Dialogue, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (!result)
{
Log.LogWarning($"Attended transfer failed.");
}
}
else
{
Log.LogWarning("There need to be two active calls before the attended transfer can occur.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
userAgent1?.Hangup();
userAgent2?.Hangup();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static async Task Main(string[] args)
{
Console.WriteLine("SIPSorcery Video Phone Command Line Demo");
Console.WriteLine("Press ctrl-c to exit.");
Log = AddConsoleLogger();
ManualResetEvent exitMRE = new ManualResetEvent(false);
ManualResetEvent waitForCallMre = new ManualResetEvent(false);
var parseResult = Parser.Default.ParseArguments <Options>(args);
_options = (parseResult as Parsed <Options>)?.Value;
if (parseResult.Tag != ParserResultType.NotParsed)
{
if (_options.ListCameras)
{
#region List webcams.
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
}
else
{
var index = 0;
foreach (var webcam in webcams)
{
Console.WriteLine($"{index}: \"{webcam.Name}\", use --cam={index}.");
index++;
}
}
#endregion
}
else if (_options.ListFormats != null)
{
#region List webcam formats.
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
}
else if (_options.ListFormats >= webcams.Count)
{
Console.WriteLine($"No webcam available for index {_options.ListFormats}.");
}
else
{
string webcamName = webcams[_options.ListFormats.Value].Name;
var formats = await WindowsVideoEndPoint.GetDeviceFrameFormats(webcamName);
Console.WriteLine($"Video frame formats for {webcamName}.");
foreach (var vidFmt in formats)
{
float vidFps = vidFmt.MediaFrameFormat.FrameRate.Numerator / vidFmt.MediaFrameFormat.FrameRate.Denominator;
string pixFmt = vidFmt.MediaFrameFormat.Subtype == WindowsVideoEndPoint.MF_I420_PIXEL_FORMAT ? "I420" : vidFmt.MediaFrameFormat.Subtype;
Console.WriteLine($"{vidFmt.Width}x{vidFmt.Height} {vidFps:0.##}fps {pixFmt}");
}
}
#endregion
}
else
{
string webcamName = null;
if (_options.WebcamIndex != null)
{
var webcams = await WindowsVideoEndPoint.GetVideoCatpureDevices();
if (webcams == null || webcams.Count == 0)
{
Console.WriteLine("No webcams were found.");
Application.Exit();
}
else if (webcams.Count < _options.WebcamIndex)
{
Console.WriteLine($"No webcam available for index {_options.WebcamIndex}.");
Application.Exit();
}
else
{
webcamName = webcams[_options.WebcamIndex.Value].Name;
Console.WriteLine($"Using webcam {webcamName}.");
}
}
_sipTransport = new SIPTransport();
if (string.IsNullOrEmpty(_options.CallDestination))
{
// We haven't been asked to place a call so we're listening.
IPAddress listenAddress = (System.Net.Sockets.Socket.OSSupportsIPv6) ? IPAddress.IPv6Any : IPAddress.Any;
var listenEndPoint = new IPEndPoint(listenAddress, SIP_PORT_DEFAULT);
try
{
SIPUDPChannel udpChannel = new SIPUDPChannel(listenEndPoint, true);
_sipTransport.AddSIPChannel(udpChannel);
}
catch (ApplicationException appExcp)
{
Console.WriteLine($"Failed to create UDP SIP channel on {listenEndPoint}, error {appExcp.Message}.");
SIPUDPChannel udpChannel = new SIPUDPChannel(new IPEndPoint(listenAddress, 0), true);
_sipTransport.AddSIPChannel(udpChannel);
}
var listeningEP = _sipTransport.GetSIPChannels().First().ListeningSIPEndPoint;
Console.WriteLine($"Listening for incoming call on {listeningEP}.");
}
EnableTraceLogs(_sipTransport);
// Open a window to display the video feed from the remote SIP party.
_form = new Form();
_form.Text = string.IsNullOrEmpty(_options.CallDestination) ? "Listener" : "Caller";
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_localVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, 0),
Visible = true
};
_remoteVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, VIDEO_FRAME_HEIGHT),
Visible = true
};
_form.Controls.Add(_localVideoPicBox);
_form.Controls.Add(_remoteVideoPicBox);
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.OnCallHungup += (dialog) => exitMRE.Set();
WindowsAudioEndPoint windowsAudioEndPoint = null;
if (!_options.NoAudio)
{
windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudioEndPoint.RestrictFormats(x => x.Codec == AudioCodecsEnum.G722);
}
MediaEndPoints mediaEndPoints = null;
if (_options.TestPattern && _options.WebcamIndex == null)
{
var testPattern = new VideoTestPatternSource(new FFmpegVideoEncoder());
var decoderSink = new DecoderVideoSink(new FFmpegVideoEncoder());
//var decoderSink = new DecoderVideoSink(new VpxVideoEncoder());
testPattern.RestrictFormats(format => format.Codec == VIDEO_CODEC);
decoderSink.RestrictFormats(format => format.Codec == VIDEO_CODEC);
mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = decoderSink,
VideoSource = testPattern,
};
}
else
{
WindowsVideoEndPoint windowsVideoEndPoint = webcamName switch
{
null => new WindowsVideoEndPoint(new FFmpegVideoEncoder()),
_ => new WindowsVideoEndPoint(new FFmpegVideoEncoder(), webcamName),
};
windowsVideoEndPoint.RestrictFormats(format => format.Codec == VIDEO_CODEC);
mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
}
mediaEndPoints.VideoSource.OnVideoSourceRawSample += (uint durationMilliseconds, int width, int height, byte[] sample, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
int stride = width * 3;
if (pixelFormat == VideoPixelFormatsEnum.I420)
{
sample = PixelConverter.I420toBGR(sample, width, height, out stride);
}
if (_localVideoPicBox.Width != width || _localVideoPicBox.Height != height)
{
Log.LogDebug($"Adjusting local video display from {_localVideoPicBox.Width}x{_localVideoPicBox.Height} to {width}x{height}.");
_localVideoPicBox.Width = width;
_localVideoPicBox.Height = height;
}
unsafe
{
fixed(byte *s = sample)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap(width, height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_localVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
waitForCallMre.Set();
exitMRE.Set();
};
if (string.IsNullOrEmpty(_options.CallDestination))
{
ActivateForm();
userAgent.OnIncomingCall += async(ua, req) =>
{
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
if (voipMediaSession.VideoLocalTrack != null)
{
voipMediaSession.VideoLocalTrack.MaximumBandwidth = MAXIMUM_VIDEO_BANDWIDTH;
}
var uas = userAgent.AcceptCall(req);
await userAgent.Answer(uas, voipMediaSession);
Console.WriteLine("Starting local video source...");
await mediaEndPoints.VideoSource.StartVideo().ConfigureAwait(false);
waitForCallMre.Set();
};
Console.WriteLine("Waiting for incoming call...");
waitForCallMre.WaitOne();
}
else
{
var voipMediaSession = new VoIPMediaSession(mediaEndPoints);
voipMediaSession.AcceptRtpFromAny = true;
if (voipMediaSession.VideoLocalTrack != null)
{
voipMediaSession.VideoLocalTrack.MaximumBandwidth = MAXIMUM_VIDEO_BANDWIDTH;
}
ActivateForm();
Console.WriteLine("Starting local video source...");
await mediaEndPoints.VideoSource.StartVideo().ConfigureAwait(false);
// Place the call and wait for the result.
Task <bool> callTask = userAgent.Call(_options.CallDestination, null, null, voipMediaSession);
callTask.Wait(CALL_TIMEOUT_SECONDS * 1000);
}
if (userAgent.IsCallActive)
{
Log.LogInformation("Call attempt successful.");
mediaEndPoints.VideoSink.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
if (_isFormActivated)
{
_form?.BeginInvoke(new Action(() =>
{
if (_form.Handle != IntPtr.Zero)
{
unsafe
{
if (_remoteVideoPicBox.Width != (int)width || _remoteVideoPicBox.Height != (int)height)
{
Log.LogDebug($"Adjusting remote video display from {_remoteVideoPicBox.Width}x{_remoteVideoPicBox.Height} to {width}x{height}.");
_remoteVideoPicBox.Width = (int)width;
_remoteVideoPicBox.Height = (int)height;
}
fixed(byte *s = bmp)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap((int)width, (int)height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_remoteVideoPicBox.Image = bmpImage;
}
}
}
}));
}
};
}
else
{
Log.LogWarning("Call attempt failed.");
Console.WriteLine("Press ctrl-c to exit.");
}
exitMRE.WaitOne();
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up.");
userAgent.Hangup();
}
Task.Delay(1000).Wait();
// Clean up.
if (_form.Handle != IntPtr.Zero)
{
_form.BeginInvoke(new Action(() => _form.Close()));
}
_sipTransport.Shutdown();
}
}
}
private static SIPEndPoint OUTBOUND_PROXY = null; // SIPEndPoint.ParseSIPEndPoint("udp:192.168.0.148:5060");
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, OUTBOUND_PROXY);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
userAgent.OnCallHungup += (dialog) => exitCts.Cancel();
var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
var callTask = userAgent.Call(DESTINATION, null, null, 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();
}
}
;
exitCts.Cancel();
};
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
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();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Video Call Demo");
Console.WriteLine("Press ctrl-c to exit.");
Log = AddConsoleLogger();
ManualResetEvent exitMRE = new ManualResetEvent(false);
_sipTransport = new SIPTransport();
EnableTraceLogs(_sipTransport);
// Open a window to display the video feed from the remote SIP party.
_form = new Form();
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_localVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, 0),
Visible = true
};
_remoteVideoPicBox = new PictureBox
{
Size = new Size(VIDEO_FRAME_WIDTH, VIDEO_FRAME_HEIGHT),
Location = new Point(0, VIDEO_FRAME_HEIGHT),
Visible = true
};
_form.Controls.Add(_localVideoPicBox);
_form.Controls.Add(_remoteVideoPicBox);
Application.EnableVisualStyles();
ThreadPool.QueueUserWorkItem(delegate { Application.Run(_form); });
ManualResetEvent formMre = new ManualResetEvent(false);
_form.Activated += (object sender, EventArgs e) => formMre.Set();
Console.WriteLine("Waiting for form activation.");
formMre.WaitOne();
_sipTransport.SIPTransportRequestReceived += OnSIPTransportRequestReceived;
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.OnCallHungup += (dialog) => exitMRE.Set();
var windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudioEndPoint.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var windowsVideoEndPoint = new WindowsVideoEndPoint(new VpxVideoEncoder());
// Fallback to a test pattern source if accessing the Windows webcam fails.
var testPattern = new VideoTestPatternSource(new VpxVideoEncoder());
MediaEndPoints mediaEndPoints = new MediaEndPoints
{
AudioSink = windowsAudioEndPoint,
AudioSource = windowsAudioEndPoint,
VideoSink = windowsVideoEndPoint,
VideoSource = windowsVideoEndPoint,
};
var voipMediaSession = new VoIPMediaSession(mediaEndPoints, testPattern);
voipMediaSession.AcceptRtpFromAny = true;
windowsVideoEndPoint.OnVideoSourceRawSample += (uint durationMilliseconds, int width, int height, byte[] sample, VideoPixelFormatsEnum pixelFormat) =>
{
_form?.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = sample)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap(width, height, width * 3, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_localVideoPicBox.Image = bmpImage;
}
}
}));
};
Console.WriteLine("Starting local video source...");
await windowsVideoEndPoint.StartVideo().ConfigureAwait(false);
// Place the call and wait for the result.
Task <bool> callTask = userAgent.Call(DESTINATION, null, null, voipMediaSession);
callTask.Wait(CALL_TIMEOUT_SECONDS * 1000);
if (callTask.Result)
{
Log.LogInformation("Call attempt successful.");
windowsVideoEndPoint.OnVideoSinkDecodedSample += (byte[] bmp, uint width, uint height, int stride, VideoPixelFormatsEnum pixelFormat) =>
{
_form?.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = bmp)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap((int)width, (int)height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_remoteVideoPicBox.Image = bmpImage;
}
}
}));
};
windowsAudioEndPoint.PauseAudio().Wait();
voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Music);
}
else
{
Log.LogWarning("Call attempt failed.");
Console.WriteLine("Press ctrl-c to exit.");
}
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
exitMRE.Set();
};
exitMRE.WaitOne();
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up.");
userAgent.Hangup();
Task.Delay(1000).Wait();
}
// Clean up.
_form.BeginInvoke(new Action(() => _form.Close()));
_sipTransport.Shutdown();
}
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();
}