/// <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);
var audioSrcOpts = new AudioOptions
{
AudioSource = AudioSourcesEnum.Microphone,
OutputDeviceIndex = m_audioOutDeviceIndex
};
var videoSrcOpts = new VideoOptions
{
VideoSource = VideoSourcesEnum.TestPattern,
SourceFile = RtpAVSession.VIDEO_TESTPATTERN,
SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND
};
MediaSession = new RtpAVSession(audioSrcOpts, videoSrcOpts);
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.InitiateCallAsync(callDescriptor, MediaSession);
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, null);
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var uas = userAgent.AcceptCall(sipRequest);
await userAgent.Answer(uas, rtpSession);
if (userAgent.IsCallActive)
{
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
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}");
}
}
/// <summary>
/// Answers an incoming SIP call.
/// </summary>
public async Task <bool> Answer()
{
if (m_pendingIncomingCall == null)
{
StatusMessage(this, $"There was no pending call available to answer.");
return(false);
}
else
{
var sipRequest = m_pendingIncomingCall.ClientTransaction.TransactionRequest;
SDP offerSDP = SDP.ParseSDPDescription(sipRequest.Body);
bool hasAudio = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
bool hasVideo = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.video && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
AudioOptions audioOpts = new AudioOptions {
AudioSource = AudioSourcesEnum.None
};
if (hasAudio)
{
audioOpts = new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
OutputDeviceIndex = m_audioOutDeviceIndex,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
};
}
VideoOptions videoOpts = new VideoOptions {
VideoSource = VideoSourcesEnum.None
};
if (hasVideo)
{
videoOpts = new VideoOptions
{
VideoSource = VideoSourcesEnum.TestPattern,
SourceFile = RtpAVSession.VIDEO_TESTPATTERN,
SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND
};
}
MediaSession = new RtpAVSession(audioOpts, videoOpts);
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
bool result = await m_userAgent.Answer(m_pendingIncomingCall, MediaSession);
m_pendingIncomingCall = null;
return(result);
}
}
static async Task DialNumber()
{
string fromHeader = (new SIPFromHeader(USERNAME, new SIPURI(USERNAME, DOMAIN, null), null)).ToString();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(USERNAME, PASSWORD, DEFAULT_CALL_DESTINATION, fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, null, null);
callDescriptor.CallId = "12028883999";
userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallRinging += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Failed: {err}, Status code: {resp?.StatusCode}");
userAgent.ClientCallAnswered += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.OnDtmfTone += (key, duration) => OnDtmfTone(userAgent, key, duration);
userAgent.OnRtpEvent += (evt, hdr) => Console.WriteLine($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
userAgent.OnCallHungup += OnHangup;
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var callResult = await userAgent.Call(callDescriptor, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
if (callResult)
{
Console.WriteLine("Enter digits one after another");
for (int i = 0; i < 11; i++)
{
var p = Console.ReadLine();
await userAgent.SendDtmf(byte.Parse(p));
}
}
Console.WriteLine("Enter ?");
Console.ReadLine();
await userAgent.SendDtmf(35);
Thread.Sleep(60000);
userAgent.Hangup();
_waveFile.Dispose();
Console.WriteLine("Hangup");
}
/// <summary>
/// Answers an incoming SIP call.
/// </summary>
public async Task Answer()
{
if (m_pendingIncomingCall == null)
{
StatusMessage(this, $"There was no pending call available to answer.");
}
else
{
var sipRequest = m_pendingIncomingCall.ClientTransaction.TransactionRequest;
SDP offerSDP = SDP.ParseSDPDescription(sipRequest.Body);
bool hasAudio = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.audio);
bool hasVideo = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.video);
AudioOptions audioOpts = new AudioOptions {
AudioSource = AudioSourcesEnum.None
};
if (hasAudio)
{
audioOpts = new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
};
}
VideoOptions videoOpts = new VideoOptions {
VideoSource = VideoSourcesEnum.None
};
if (hasVideo)
{
videoOpts = new VideoOptions
{
VideoSource = VideoSourcesEnum.TestPattern,
SourceFile = RtpAVSession.VIDEO_TESTPATTERN,
SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND
};
}
MediaSession = new RtpAVSession(sipRequest.RemoteSIPEndPoint.Address.AddressFamily, audioOpts, videoOpts);
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.Answer(m_pendingIncomingCall, MediaSession);
m_pendingIncomingCall = null;
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.Microphone,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"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(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
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("localhost.pfx");
// 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.
RtpAVSession 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)
{
SIPSorcery.Sys.Log.Logger.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.HasMediaFormat((int)SDPMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
rtpSession = new RtpAVSession(
new AudioOptions {
AudioSource = AudioSourcesEnum.Music, SourceFile = executableDir + "/" + AUDIO_FILE_PCMU
}, null);
rtpSession.setRemoteDescription(new RTCSessionDescription {
type = RTCSdpType.offer, sdp = offerSdp
});
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
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, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.CloseSession(null);
};
rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = await rtpSession.createAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.CloseSession(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)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.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?.CloseSession(null);
rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
_currentDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
RtpAVSession rtpAVSession = 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.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
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);
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await userAgent.Answer(incomingCall, rtpAVSession);
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);
}
};
// 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)
{
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, rtpAVSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
userAgent.TakeOffHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
userAgent.PutOnHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions
{
AudioSource = AudioSourcesEnum.Music,
SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>
{
{ SDPMediaFormatsEnum.PCMU, _currentDir + "/" + AUDIO_FILE_PCMU }
}
}, null);
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"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)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpAVSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"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();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
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.
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, rtpSession);
// 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();
SIPSorcery.Net.DNSManager.Stop();
}
static async Task 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;
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);
// Get the IP address the RTP will be sent from. While we can listen on IPAddress.Any | IPv6Any
// we can't put 0.0.0.0 or [::0] in the SDP or the callee will treat our RTP stream as inactive.
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var rtpSession = new RtpAVSession(dstAddress.AddressFamily, new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, null);
var offerSDP = await rtpSession.createOffer(new RTCOfferOptions { RemoteSignallingAddress = dstAddress });
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{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($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
rtpSession.setRemoteDescription(new RTCSessionDescription {
type = RTCSdpType.answer, sdp = SDP.ParseSDPDescription(resp.Body)
});
rtpSession.Start();
}
else
{
Log.LogWarning($"{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);
uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
rtpSession.CloseSession(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, err, resp) =>
{
Console.WriteLine($"Call failed {err}");
_waveFile?.Close();
};
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.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, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
//private delegate void MediaSampleReadyDelegate(SDPMediaTypesEnum mediaType, uint duration, byte[] sample);
//private static event MediaSampleReadyDelegate OnMediaFromSIPSampleReady;
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery SIP to WebRTC example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Start web socket.
Console.WriteLine("Starting web socket server...");
_webSocketServer = new WebSocketServer(IPAddress.Any, WEBSOCKET_PORT, true);
_webSocketServer.SslConfiguration.ServerCertificate = new X509Certificate2(WEBSOCKET_CERTIFICATE_PATH);
_webSocketServer.SslConfiguration.CheckCertificateRevocation = false;
//_webSocketServer.Log.Level = WebSocketSharp.LogLevel.Debug;
_webSocketServer.AddWebSocketService <SDPExchange>("/", (sdpExchanger) =>
{
sdpExchanger.WebSocketOpened += SendSDPOffer;
sdpExchanger.SDPAnswerReceived += SDPAnswerReceived;
});
_webSocketServer.Start();
Console.WriteLine($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
RtpAVSession rtpAVSession = null;
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
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($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await userAgent.Answer(incomingCall, rtpAVSession);
rtpAVSession.OnRtpPacketReceived += (mediaType, rtpPacket) => ForwardMedia(mediaType, rtpPacket);
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpAVSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"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();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
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.
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);
var rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await activeAgent.Answer(incomingCall, rtpAVSession);
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($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
// Place an outgoing call using the first free user agent.
SIPUserAgent freeAgent = (!userAgent1.IsCallActive) ? userAgent1 : (!userAgent2.IsCallActive) ? userAgent2 : null;
if (freeAgent != null)
{
var rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await freeAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, rtpAVSession);
Log.LogInformation($"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($"Attended transfer failed.");
}
}
else
{
Log.LogWarning("There need to be two active calls before the attended transfer can occur.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"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...");
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 void Main()
{
Console.WriteLine("SIPSorcery Getting Started Video Call Demo");
Console.WriteLine("Press ctrl-c to exit.");
AddConsoleLogger();
_sipTransport = new SIPTransport();
EnableTraceLogs(_sipTransport);
// Open a Window to display the video feed from the WebRTC peer.
_form = new Form();
_form.AutoSize = true;
_form.BackgroundImageLayout = ImageLayout.Center;
_picBox = new PictureBox
{
Size = new Size(640, 480),
Location = new Point(0, 0),
Visible = true
};
_form.Controls.Add(_picBox);
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);
var audioSrcOpts = new AudioOptions
{
AudioSource = AudioSourcesEnum.Music,
SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>
{
{ SDPMediaFormatsEnum.PCMU, executableDir + "/" + AUDIO_FILE_PCMU }
}
};
var videoSrcOpts = new VideoOptions {
VideoSource = VideoSourcesEnum.TestPattern, SourceFile = executableDir + "/" + VIDEO_TEST_PATTERN_FILE
};
var rtpSession = new RtpAVSession(audioSrcOpts, videoSrcOpts);
// Place the call and wait for the result.
Task <bool> callTask = userAgent.Call(DESTINATION, null, null, rtpSession);
callTask.Wait(CALL_IMTEOUT_SECONDS * 1000);
ManualResetEvent exitMRE = new ManualResetEvent(false);
if (callTask.Result)
{
Log.LogInformation("Call attempt successful.");
rtpSession.OnVideoSampleReady += (byte[] sample, uint width, uint height, int stride) =>
{
_picBox.BeginInvoke(new Action(() =>
{
unsafe
{
fixed(byte *s = sample)
{
System.Drawing.Bitmap bmpImage = new System.Drawing.Bitmap((int)width, (int)height, stride, System.Drawing.Imaging.PixelFormat.Format24bppRgb, (IntPtr)s);
_picBox.Image = bmpImage;
}
}
}));
};
}
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();
}
