public SIPFlow(SIPTransport sipTransport, List<ActionDiagram> actionDiagrams, List<DecisionDiagram> decisionDiagrams)
: base(actionDiagrams, decisionDiagrams)
{
m_sipTransport = sipTransport;
pythonHelper = new PythonHelper(m_pythonEngine, m_sipTransport);
pythonHelper.LogEvent += new SIPFlowLogDelegate(pythonHelper_LogEvent);
//m_sipTransport.SIPTransportResponseReceived += new SIPTransportResponseReceivedDelegate(pythonHelper.SIPTransportResponseReceived);
//m_pythonEngine.Import("BlueFace.VoIP.Net.SIP.*");
m_pythonEngine.Execute("import clr");
//m_pythonEngine.Execute("clr.AddReference('BlueFace.VoIP.Net')");
//m_pythonEngine.Execute("from BlueFace.VoIP.Net.SIP import *");
//m_pythonEngine.Execute("clr.AddReference('siggui')");
//m_pythonEngine.Execute("from BlueFace.Net.SignallingGUI import *");
m_pythonEngine.Globals["pythonHelper"] = pythonHelper;
//m_pythonEngine.Globals["sipRequest"] = sipRequest;
//SIPTransaction transaction = new SIPTransaction(sipRequest);
//SIPTransport.SendSIPReliable(transaction);
m_sipFlowDebugStream = new MemoryStream();
m_debugStreamReader = new StreamReader(m_sipFlowDebugStream);
m_pythonEngine.SetStandardOutput(m_sipFlowDebugStream);
}
public SIPNonInviteServerUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
string sipUsername,
string sipDomain,
SIPCallDirection callDirection,
SIPAssetGetDelegate<SIPAccount> getSIPAccount,
SIPAuthenticateRequestDelegate sipAuthenticateRequest,
SIPMonitorLogDelegate logDelegate,
SIPNonInviteTransaction transaction)
{
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_sipUsername = sipUsername;
m_sipDomain = sipDomain;
m_sipCallDirection = callDirection;
GetSIPAccount_External = getSIPAccount;
SIPAuthenticateRequest_External = sipAuthenticateRequest;
Log_External = logDelegate ?? Log_External;
m_transaction = transaction;
m_transaction.TransactionTraceMessage += TransactionTraceMessage;
//m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
//m_uasTransaction.UASInviteTransactionCancelled += UASTransactionCancelled;
//m_uasTransaction.TransactionRemoved += new SIPTransactionRemovedDelegate(UASTransaction_TransactionRemoved);
//m_uasTransaction.TransactionStateChanged += (t) => { logger.Debug("Transaction state change to " + t.TransactionState + ", uri=" + t.TransactionRequestURI.ToString() + "."); };
}
public SIPReferServerUserAgent(SIPTransport sipTransport, SIPMonitorLogDelegate logDelegate, SIPNonInviteTransaction sipTransaction)
{
m_sipTransport = sipTransport;
Log_External = logDelegate;
m_sipTransaction = sipTransaction;
m_sipTransaction.TransactionTraceMessage += TransactionTraceMessage;
m_sipTransaction.NonInviteTransactionTimedOut += ClientTimedOut;
// If external logging is not required assign an empty handler to stop null reference exceptions.
if (Log_External == null)
{
Log_External = (e) => { };
}
var referTo = SIPURI.ParseSIPURI(m_sipTransaction.TransactionRequest.Header.ReferTo);
var replacesParameter = SIPReplacesParameter.Parse(referTo.Headers.Get("Replaces"));
ReplacedCall = new ReplacesCallDescriptor();
ReplacedCall.CallId = replacesParameter.CallID;
ReplacedCall.FromTag = replacesParameter.FromTag;
ReplacedCall.ToTag = replacesParameter.ToTag;
ReferToUri = referTo.CopyOf();
ReferToUri.Headers.RemoveAll();
}
public SIPReferClientUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
string owner,
string adminMemberId,
SIPMonitorLogDelegate logDelegate)
{
m_sipTransport = sipTransport;
m_outboundProxy = (outboundProxy != null) ? SIPEndPoint.ParseSIPEndPoint(outboundProxy.ToString()) : null;
Owner = owner;
AdminMemberId = adminMemberId;
Log_External = logDelegate;
// If external logging is not required assign an empty handler to stop null reference exceptions.
if (Log_External == null)
{
Log_External = (e) => { };
}
}
public SIPTransferServerUserAgent(
SIPMonitorLogDelegate logDelegate,
BlindTransferDelegate blindTransfer,
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
SIPDialogue dialogueToReplace,
SIPDialogue oppositeDialogue,
string callDestination,
string owner,
string adminID)
{
Log_External = logDelegate;
BlindTransfer_External = blindTransfer;
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_dialogueToReplace = dialogueToReplace;
m_oppositeDialogue = oppositeDialogue;
m_callDestination = callDestination;
m_owner = owner;
m_adminID = adminID;
m_dummyRequest = CreateDummyRequest(m_dialogueToReplace, m_callDestination);
}
/// <summary>
/// Initialises the SIP transport layer.
/// </summary>
public async Task InitialiseSIP()
{
if (_isInitialised == false)
{
await Task.Run(() =>
{
_isInitialised = true;
// Configure the SIP transport layer.
SIPTransport = new SIPTransport();
bool sipChannelAdded = false;
if (m_sipSocketsNode != null)
{
// Set up the SIP channels based on the app.config file.
List <SIPChannel> sipChannels = SIPTransportConfig.ParseSIPChannelsNode(m_sipSocketsNode);
if (sipChannels?.Count > 0)
{
SIPTransport.AddSIPChannel(sipChannels);
sipChannelAdded = true;
}
}
if (sipChannelAdded == false)
{
// Use default options to set up a SIP channel.
SIPUDPChannel udpChannel = null;
try
{
udpChannel = new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_DEFAULT_PORT));
}
catch (ApplicationException bindExcp)
{
logger.LogWarning($"Socket exception attempting to bind UDP channel to port {SIP_DEFAULT_PORT}, will use random port. {bindExcp.Message}.");
udpChannel = new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0));
}
SIPTCPChannel tcpChannel = null;
try
{
tcpChannel = new SIPTCPChannel(new IPEndPoint(IPAddress.Any, udpChannel.Port));
}
catch (SocketException bindExcp)
{
logger.LogWarning($"Socket exception attempting to bind TCP channel to port {udpChannel.Port}, will use random port. {bindExcp.Message}.");
tcpChannel = new SIPTCPChannel(new IPEndPoint(IPAddress.Any, 0));
}
SIPTransport.AddSIPChannel(new List <SIPChannel> {
udpChannel, tcpChannel
});
}
});
// Wire up the transport layer so incoming SIP requests have somewhere to go.
SIPTransport.SIPTransportRequestReceived += SIPTransportRequestReceived;
// Log all SIP packets received to a log file.
SIPTransport.SIPRequestInTraceEvent += SIPRequestInTraceEvent;
SIPTransport.SIPRequestOutTraceEvent += SIPRequestOutTraceEvent;
SIPTransport.SIPResponseInTraceEvent += SIPResponseInTraceEvent;
SIPTransport.SIPResponseOutTraceEvent += SIPResponseOutTraceEvent;
}
}
public void AddRegisterRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest registerRequest)
{
try
{
if (registerRequest.Method != SIPMethodsEnum.REGISTER)
{
SIPResponse notSupportedResponse = GetErrorResponse(registerRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, "Registration requests only");
m_sipTransport.SendResponse(notSupportedResponse);
}
else
{
SIPSorceryPerformanceMonitor.IncrementCounter(SIPSorceryPerformanceMonitor.REGISTRAR_REGISTRATION_REQUESTS_PER_SECOND);
int requestedExpiry = GetRequestedExpiry(registerRequest);
if (registerRequest.Header.To == null)
{
logger.Debug("Bad register request, no To header from " + remoteEndPoint + ".");
SIPResponse badReqResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing To header");
m_sipTransport.SendResponse(badReqResponse);
}
else if (registerRequest.Header.To.ToURI.User.IsNullOrBlank())
{
logger.Debug("Bad register request, no To user from " + remoteEndPoint + ".");
SIPResponse badReqResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing username on To header");
m_sipTransport.SendResponse(badReqResponse);
}
else if (registerRequest.Header.Contact == null || registerRequest.Header.Contact.Count == 0)
{
logger.Debug("Bad register request, no Contact header from " + remoteEndPoint + ".");
SIPResponse badReqResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing Contact header");
m_sipTransport.SendResponse(badReqResponse);
}
else if (requestedExpiry > 0 && requestedExpiry < m_minimumBindingExpiry)
{
logger.Debug("Bad register request, no expiry of " + requestedExpiry + " to small from " + remoteEndPoint + ".");
SIPResponse tooFrequentResponse = GetErrorResponse(registerRequest, SIPResponseStatusCodesEnum.IntervalTooBrief, null);
tooFrequentResponse.Header.MinExpires = m_minimumBindingExpiry;
m_sipTransport.SendResponse(tooFrequentResponse);
}
else
{
if (m_registerQueue.Count < MAX_REGISTER_QUEUE_SIZE)
{
SIPNonInviteTransaction registrarTransaction = m_sipTransport.CreateNonInviteTransaction(registerRequest, remoteEndPoint, localSIPEndPoint, null);
lock (m_registerQueue)
{
m_registerQueue.Enqueue(registrarTransaction);
}
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingInProgress, "Register queued for " + registerRequest.Header.To.ToURI.ToString() + ".", null));
}
else
{
logger.Error("Register queue exceeded max queue size " + MAX_REGISTER_QUEUE_SIZE + ", overloaded response sent.");
SIPResponse overloadedResponse = SIPTransport.GetResponse(registerRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, "Registrar overloaded, please try again shortly");
m_sipTransport.SendResponse(overloadedResponse);
}
m_registerARE.Set();
}
}
}
catch (Exception excp)
{
logger.Error("Exception AddRegisterRequest (" + remoteEndPoint.ToString() + "). " + excp.Message);
}
}
/// <summary>
/// Helper method for dynamic proxy runtime script.
/// </summary>
/// <param name="responseCode"></param>
/// <param name="localEndPoint"></param>
/// <param name="remoteEndPoint"></param>
/// <param name="sipRequest"></param>
public void Respond(SIPRequest sipRequest, SIPResponseStatusCodesEnum responseCode, string reasonPhrase)
{
SIPResponse response = SIPTransport.GetResponse(sipRequest, responseCode, reasonPhrase);
m_sipTransport.SendResponse(response);
}
static void Main()
{
Console.WriteLine("SIPSorcery call hold 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.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
// Get the default speaker.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
WaveInEvent waveInEvent = GetAudioInputDevice();
RTPMediaSession RtpMediaSession = 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.ClientCallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
exitCts.Cancel();
};
userAgent.ClientCallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
PlayRemoteMedia(RtpMediaSession, audioOutProvider);
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
hasCallFailed = true;
exitCts.Cancel();
}
};
userAgent.OnCallHungup += () =>
{
Log.LogInformation($"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);
RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
await userAgent.Answer(incomingCall, RtpMediaSession);
PlayRemoteMedia(RtpMediaSession, audioOutProvider);
waveInEvent.StartRecording();
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);
}
};
// Wire up the RTP send session to the audio output device.
uint rtpSendTimestamp = 0;
waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) =>
{
byte[] sample = new byte[args.Buffer.Length / 2];
int sampleIndex = 0;
for (int index = 0; index < args.BytesRecorded; index += 2)
{
var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index));
sample[sampleIndex++] = ulawByte;
}
if (RtpMediaSession != null)
{
RtpMediaSession.SendAudioFrame(rtpSendTimestamp, sample);
rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds);
}
};
// 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)
{
RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork);
RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
var callDescriptor = GetCallDescriptor(DEFAULT_DESTINATION_SIP_URI);
await userAgent.InitiateCall(callDescriptor, RtpMediaSession);
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (RtpMediaSession.LocalOnHold)
{
Log.LogInformation("Taking the remote call party off hold.");
RtpMediaSession.TakeOffHold();
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
RtpMediaSession.PutOnHold();
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 't')
{
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...");
RtpMediaSession?.Close();
waveInEvent?.StopRecording();
audioOutEvent?.Stop();
if (!isCallHungup && userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {userAgent?.CallDescriptor?.To}.");
userAgent.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();
}
#endregion
}
/// <summary>
/// Handler for processing incoming SIP requests.
/// </summary>
/// <param name="localSIPEndPoint">The end point the request was received on.</param>
/// <param name="remoteEndPoint">The end point the request came from.</param>
/// <param name="sipRequest">The SIP request received.</param>
private void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (m_uac != null && m_uac.SIPDialogue != null && sipRequest.Header.CallId == m_uac.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup by remote end.");
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallFinished();
}
else if (m_uas != null && m_uas.SIPDialogue != null && sipRequest.Header.CallId == m_uas.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup.");
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallFinished();
}
else
{
logger.Debug("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
StatusMessage("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
m_uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
m_uas.CallCancelled += UASCallCancelled;
m_uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
m_uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
IncomingCall();
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
StatusMessage("Call was cancelled by remote end.");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
logger.Debug("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
CallFinished();
}
else
{
logger.Debug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
//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($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
ManualResetEvent exitMre = new ManualResetEvent(false);
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0], out callUri))
{
Log.LogWarning($"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;
var localOfferAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var audioOptions = new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.PCMU
},
OutputDeviceIndex = AudioOptions.DEFAULT_OUTPUTDEVICE_INDEX
};
var rtpSession = new RtpAVSession(audioOptions, null);
var offerSDP = rtpSession.CreateOffer(localOfferAddress);
// 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}.");
if (resp.Status == SIPResponseStatusCodesEnum.SessionProgress)
{
rtpSession.Start();
}
};
uac.CallFailed += (uac, err, resp) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (iuac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var result = rtpSession.SetRemoteDescription(SDP.ParseSDPDescription(resp.Body));
if (result == SetDescriptionResultEnum.OK)
{
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);
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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
private static SIPEndPoint OUTBOUND_PROXY = null; // SIPEndPoint.ParseSIPEndPoint("udp:192.168.0.148:5060");
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, OUTBOUND_PROXY);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"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();
}
//private SIPAccount m_destinationSIPAccount;
public SIPB2BUserAgent(
SIPMonitorLogDelegate logDelegate,
QueueNewCallDelegate queueCall,
SIPTransport sipTranpsort,
string uacOwner,
string uacAdminMemberId
)
{
Log_External = logDelegate;
QueueNewCall_External = queueCall;
m_sipTransport = sipTranpsort;
m_uacOwner = uacOwner;
m_uacAdminMemberId = uacAdminMemberId;
}
public PythonHelper(PythonEngine pythonEngine, SIPTransport sipTransport)
{
m_sipTransport = sipTransport;
logger.Debug("PythonHelper SIP Transport local socket is " + m_sipTransport.GetDefaultTransportContact(SIPProtocolsEnum.UDP) + ".");
}
public bool AuthenticateCall()
{
m_isAuthenticated = false;
try
{
if (SIPAuthenticateRequest_External == null)
{
// No point trying to authenticate if we haven't been given an authentication delegate.
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
else if (GetSIPAccount_External == null)
{
// No point trying to authenticate if we haven't been given a delegate to load the SIP account.
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
else
{
m_sipAccount = GetSIPAccount_External(s => s.SIPUsername == m_sipUsername && s.SIPDomain == m_sipDomain);
if (m_sipAccount == null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Rejecting authentication required call for " + m_sipUsername + "@" + m_sipDomain + ", SIP account not found.", null));
Reject(SIPResponseStatusCodesEnum.Forbidden, null, null);
}
else
{
SIPRequest sipRequest = m_uasTransaction.TransactionRequest;
SIPEndPoint localSIPEndPoint = (!sipRequest.Header.ProxyReceivedOn.IsNullOrBlank()) ? SIPEndPoint.ParseSIPEndPoint(sipRequest.Header.ProxyReceivedOn) : sipRequest.LocalSIPEndPoint;
SIPEndPoint remoteEndPoint = (!sipRequest.Header.ProxyReceivedFrom.IsNullOrBlank()) ? SIPEndPoint.ParseSIPEndPoint(sipRequest.Header.ProxyReceivedFrom) : sipRequest.RemoteSIPEndPoint;
SIPRequestAuthenticationResult authenticationResult = SIPAuthenticateRequest_External(localSIPEndPoint, remoteEndPoint, sipRequest, m_sipAccount, Log_External);
if (authenticationResult.Authenticated)
{
if (authenticationResult.WasAuthenticatedByIP)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "New call from " + remoteEndPoint.ToString() + " successfully authenticated by IP address.", m_sipAccount.Owner));
}
else
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "New call from " + remoteEndPoint.ToString() + " successfully authenticated by digest.", m_sipAccount.Owner));
}
SetOwner(m_sipAccount.Owner, m_sipAccount.AdminMemberId);
m_isAuthenticated = true;
}
else
{
// Send authorisation failure or required response
SIPResponse authReqdResponse = SIPTransport.GetResponse(sipRequest, authenticationResult.ErrorResponse, null);
authReqdResponse.Header.AuthenticationHeader = authenticationResult.AuthenticationRequiredHeader;
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Call not authenticated for " + m_sipUsername + "@" + m_sipDomain + ", responding with " + authenticationResult.ErrorResponse + ".", null));
m_uasTransaction.SendFinalResponse(authReqdResponse);
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception SIPServerUserAgent AuthenticateCall. " + excp.Message);
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
return(m_isAuthenticated);
}
/// <summary>
/// Enable detailed SIP log messages.
/// </summary>
private static void EnableTraceLogs(SIPTransport sipTransport, bool fullSIP)
{
sipTransport.SIPRequestInTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request received: {localEP}<-{remoteEP}");
if (!fullSIP)
{
Log.LogDebug(req.StatusLine);
}
else
{
Log.LogDebug(req.ToString());
}
};
sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request sent: {localEP}->{remoteEP}");
if (!fullSIP)
{
Log.LogDebug(req.StatusLine);
}
else
{
Log.LogDebug(req.ToString());
}
};
sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response received: {localEP}<-{remoteEP}");
if (!fullSIP)
{
Log.LogDebug(resp.ShortDescription);
}
else
{
Log.LogDebug(resp.ToString());
}
};
sipTransport.SIPResponseOutTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response sent: {localEP}->{remoteEP}");
if (!fullSIP)
{
Log.LogDebug(resp.ShortDescription);
}
else
{
Log.LogDebug(resp.ToString());
}
};
sipTransport.SIPRequestRetransmitTraceEvent += (tx, req, count) =>
{
Log.LogDebug($"Request retransmit {count} for request {req.StatusLine}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
sipTransport.SIPResponseRetransmitTraceEvent += (tx, resp, count) =>
{
Log.LogDebug($"Response retransmit {count} for response {resp.ShortDescription}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
}
static void Main()
{
Console.WriteLine("SIPSorcery Attended Transfer Demo: Transfer Target");
Console.WriteLine("Waiting for incoming call from Transferor.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(_sipTransport);
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => Log.LogDebug("Call hungup by remote party.");
userAgent.OnIncomingCall += async(ua, req) =>
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs, null, _rtpPort);
_rtpPort += 2;
rtpAudioSession.OnReceiveReport += RtpSession_OnReceiveReport;
//rtpAudioSession.OnSendReport += RtpSession_OnSendReport;
var uas = ua.AcceptCall(req);
bool answerResult = await ua.Answer(uas, rtpAudioSession);
Log.LogDebug($"Answer incoming call result {answerResult}.");
_ = Task.Run(async() =>
{
await Task.Delay(1000);
Log.LogDebug($"Sending DTMF sequence {string.Join("", DTMF_SEQUENCEFOR_TRANSFEROR.Select(x => x))}.");
foreach (byte dtmf in DTMF_SEQUENCEFOR_TRANSFEROR)
{
Log.LogDebug($"Sending DTMF tone to transferor {dtmf}.");
await ua.SendDtmf(dtmf);
}
});
};
userAgent.OnDtmfTone += (key, duration) => Log.LogInformation($"Received DTMF tone {key}.");
Task.Run(() => OnKeyPress(userAgent, exitCts));
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
//userAgent?.Hangup();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
SIPSorcery.Net.DNSManager.Stop();
if (_sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
_sipTransport.Shutdown();
}
#endregion
}
public SIPClientUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
string owner,
string adminMemberId,
SIPMonitorLogDelegate logDelegate,
RtccGetCustomerDelegate rtccGetCustomer,
RtccGetRateDelegate rtccGetRate,
RtccGetBalanceDelegate rtccGetBalance,
RtccReserveInitialCreditDelegate rtccReserveInitialCredit,
RtccUpdateCdrDelegate rtccUpdateCdr
)
: this(sipTransport, outboundProxy, owner, adminMemberId, logDelegate)
{
RtccGetCustomer_External = rtccGetCustomer;
RtccGetRate_External = rtccGetRate;
RtccGetBalance_External = rtccGetBalance;
RtccReserveInitialCredit_External = rtccReserveInitialCredit;
RtccUpdateCdr_External = rtccUpdateCdr;
}
static void Main()
{
Console.WriteLine("SIPSorcery registration user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var sipChannel = new SIPUDPChannel(IPAddress.Any, 0);
sipTransport.AddSIPChannel(sipChannel);
//EnableTraceLogs(sipTransport);
// Create a client user agent to maintain a periodic registration with a SIP server.
var regUserAgent = new SIPRegistrationUserAgent(
sipTransport,
"softphonesample",
"password",
"sipsorcery.com",
120);
int successCounter = 0;
ManualResetEvent taskCompleteMre = new ManualResetEvent(false);
// Event handlers for the different stages of the registration.
regUserAgent.RegistrationFailed += (uri, err) => SIPSorcery.Sys.Log.Logger.LogError($"{uri.ToString()}: {err}");
regUserAgent.RegistrationTemporaryFailure += (uri, msg) => SIPSorcery.Sys.Log.Logger.LogWarning($"{uri.ToString()}: {msg}");
regUserAgent.RegistrationRemoved += (uri) => SIPSorcery.Sys.Log.Logger.LogError($"{uri.ToString()} registration failed.");
regUserAgent.RegistrationSuccessful += (uri) =>
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{uri.ToString()} registration succeeded.");
Interlocked.Increment(ref successCounter);
SIPSorcery.Sys.Log.Logger.LogInformation($"Successful registrations {successCounter} of {SUCCESS_REGISTRATION_COUNT}.");
if (successCounter == SUCCESS_REGISTRATION_COUNT)
{
taskCompleteMre.Set();
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
taskCompleteMre.Set();
};
// Start the thread to perform the initial registration and then periodically resend it.
regUserAgent.Start();
taskCompleteMre.WaitOne();
regUserAgent.Stop();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
SIPSorcery.Net.DNSManager.Stop();
}
private static readonly int RTP_REPORTING_PERIOD_SECONDS = 5; // Period at which to write RTP stats.
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource cts = new CancellationTokenSource();
bool isCallHungup = false;
bool hasCallFailed = false;
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Loopback, port)));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Loopback, port)));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(LocalIPConfig.GetDefaultIPv4Address(), port)));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(LocalIPConfig.GetDefaultIPv6Address(), port)));
// Select the IP address to use for RTP based on the destination SIP URI.
SIPURI callURI = SIPURI.ParseSIPURIRelaxed(DESTINATION_SIP_URI);
var endPointForCall = callURI.ToSIPEndPoint() == null?sipTransport.GetDefaultSIPEndPoint(callURI.Protocol) : sipTransport.GetDefaultSIPEndPoint(callURI.ToSIPEndPoint());
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(endPointForCall.Address, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, 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) =>
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
SIPSorcery.Sys.Log.Logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
SIPSorcery.Sys.Log.Logger.LogDebug(resp.ToString());
IPEndPoint remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
SIPSorcery.Sys.Log.Logger.LogDebug($"Sending initial RTP packet to remote RTP socket {remoteRtpEndPoint}.");
// Send a dummy packet to open the NAT session on the RTP path.
rtpSendSession.SendAudioFrame(rtpSocket, remoteRtpEndPoint, 0, new byte[] { 0x00 });
}
else
{
SIPSorcery.Sys.Log.Logger.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 += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
cts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => SendRecvRtp(rtpSocket, rtpSendSession, cts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
DESTINATION_SIP_URI,
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += async delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
cts.Cancel();
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to clean up...");
await Task.Delay(1000);
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
public async Task IncomingCallNoSdpWithACKUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
var dummySep = SIPEndPoint.ParseSIPEndPoint("udp:127.0.0.1:5060");
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = @"INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:51200;branch=z9hG4bKbeed9b0cde8d43cc8a2aae91526b6a1d;rport
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=GCLNRILCDU
Call-ID: 7265e19f53a146a1bacdf4f4f8ea70b2
CSeq: 1 INVITE
Contact: <sip:127.0.0.1:51200>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Length: 0
Content-Type: application/sdp" + m_CRLF + m_CRLF;
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
var mediaSession = CreateMediaSession();
_ = Task.Run(() =>
{
Task.Delay(2000).Wait();
string ackReqStr = @"ACK sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:51200;branch=z9hG4bK76dfb1480ea14f778bd24afed1c8ded0;rport
To: <sip:[email protected]>;tag=YWPNZPMLPB
From: <sip:[email protected]>;tag=GCLNRILCDU
Call-ID: 7265e19f53a146a1bacdf4f4f8ea70b2
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 160
v=0
o=- 67424 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 16976 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv" + m_CRLF + m_CRLF;
uas.ClientTransaction.ACKReceived(dummySep, dummySep, SIPRequest.ParseSIPRequest(ackReqStr));
});
await userAgent.Answer(uas, mediaSession);
Assert.True(userAgent.IsCallActive);
}
private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
var rtpJob = (from job in m_rtpJobs.Values where job.UAS.CallRequest.Header.CallId == sipRequest.Header.CallId select job).FirstOrDefault();
if (rtpJob != null)
{
rtpJob.Stop();
// Call has been hungup by remote end.
Console.WriteLine("Call hungup by client: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
Publish(rtpJob.QueueName, "BYE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
//Console.WriteLine("Request Received " + localSIPEndPoint + "<-" + remoteEndPoint + "\n" + sipRequest.ToString());
//m_uas.SIPDialogue.Hangup(m_sipTransport, null);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
}
else
{
Console.WriteLine("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".\n");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Console.WriteLine("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
Publish(sipRequest.URI.User, "INVITE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
Console.WriteLine(sipRequest.Body);
SIPPacketMangler.MangleSIPRequest(SIPMonitorServerTypesEnum.Unknown, sipRequest, null, LogTraceMessage);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
var uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, LogTraceMessage, uasTransaction);
uas.CallCancelled += UASCallCancelled;
RTPDiagnosticsJob rtpJob = new RTPDiagnosticsJob(m_rtpListenIPAddress, m_publicIPAddress, uas, sipRequest);
string sdpAddress = SDP.GetSDPRTPEndPoint(sipRequest.Body).Address.ToString();
// Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.
IPEndPoint expectedRTPEndPoint = new IPEndPoint(rtpJob.RemoteRTPEndPoint.Address, rtpJob.RemoteRTPEndPoint.Port);
if (IPSocket.IsPrivateAddress(rtpJob.RemoteRTPEndPoint.Address.ToString()))
{
expectedRTPEndPoint.Address = remoteEndPoint.Address;
}
Publish(sipRequest.URI.User, "Advertised RTP remote socket " + rtpJob.RemoteRTPEndPoint + ", expecting from " + expectedRTPEndPoint + ".");
m_rtpJobs.Add(rtpJob.RTPListenEndPoint.Port, rtpJob);
//ThreadPool.QueueUserWorkItem(delegate { StartRTPListener(rtpJob); });
Console.WriteLine(rtpJob.LocalSDP.ToString());
uas.Answer("application/sdp", rtpJob.LocalSDP.ToString(), CallProperties.CreateNewTag(), null, SIPDialogueTransferModesEnum.NotAllowed);
var hangupTimer = new Timer(delegate
{
if (!rtpJob.StopJob)
{
if (uas != null && uas.SIPDialogue != null)
{
if (rtpJob.RTPPacketReceived && !rtpJob.ErrorOnRTPSend)
{
Publish(sipRequest.URI.User, "Test completed. There were no RTP send or receive errors.");
}
else if (!rtpJob.RTPPacketReceived)
{
Publish(sipRequest.URI.User, "Test completed. An error was identified, no RTP packets were received.");
}
else
{
Publish(sipRequest.URI.User, "Test completed. An error was identified, there was a problem when attempting to send an RTP packet.");
}
rtpJob.Stop();
uas.SIPDialogue.Hangup(m_sipTransport, null);
}
}
}, null, HANGUP_TIMEOUT, Timeout.Infinite);
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
Console.WriteLine("Matching CANCEL request received " + sipRequest.URI.ToString() + ".\n");
Publish(sipRequest.URI.User, "CANCEL request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
Console.WriteLine("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".\n");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else
{
Console.WriteLine("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.\n");
Publish(sipRequest.URI.User, sipRequest.Method + " request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
private void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
switch (sipRequest.Method)
{
case SIPMethodsEnum.OPTIONS:
SIPNonInviteTransaction optionsTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse optionsResponse = SipHelper.WG67ResponseNormalize(
SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null));
optionsTransaction.SendFinalResponse(optionsResponse);
OptionsReceived?.Invoke(remoteEndPoint.ToString());
break;
case SIPMethodsEnum.SUBSCRIBE:
m_sip_notifier.AddSubscribeRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
break;
case SIPMethodsEnum.PUBLISH:
default:
throw new NotImplementedException();
}
}
catch (NotImplementedException)
{
_logger.From().Debug(sipRequest.Method + " request processing not implemented for " + sipRequest.URI.ToParameterlessString() + " from " + remoteEndPoint + ".");
SIPNonInviteTransaction notImplTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse notImplResponse = SipHelper.WG67ResponseNormalize(SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotImplemented, null));
notImplTransaction.SendFinalResponse(notImplResponse);
}
catch (Exception x)
{
_logger.From().Debug($"SIPTransportRequestReceived Exception {x.Message}", x);
}
}
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(AddressFamily.InterNetwork, new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, 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();
}
/// <summary>
/// 计时器初始化
/// </summary>
/// <param name="request">sip响应</param>
/// <param name="trans">sip传输</param>
public TaskTiming(SIPRequest request, SIPTransport trans)
{
_request = request;
_trans = trans;
_timeSend = new Timer();
}
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);
userAgent.OnCallHungup += (dialog) => exitMRE.Set();
var windowsAudioEndPoint = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudioEndPoint.RestrictFormats(format => format.Codec == AudioCodecsEnum.PCMU);
var windowsVideoEndPoint = new WindowsVideoEndPoint();
//windowsVideoEndPoint.OnVideoSourceError += (err) =>
//{
// Log.LogError($"Video source error. {err}");
// if (userAgent.IsCallActive)
// {
// userAgent.Hangup();
// }
//};
// Fallback to a test pattern source if accessing the Windows webcam fails.
var testPattern = new VideoTestPatternSource(new VideoEncoder());
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((int)width, (int)height, (int)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();
}
public override void Initialize(SIPAuthenticateRequestDelegate sipRequestAuthenticator, SIPAccount account, SIPTransport transport)
{
PlatformType = account.PlatformType;
LocalSIPId = account.LocalID;
LocalEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, account.LocalIP, account.LocalPort);
RemoteSIPId = account.RemoteID;
RemoteEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, account.RemoteIP, account.RemotePort);
Transport = transport;
if (PlatformType == Cores.PlatformType.UpPlatform)
{
Start();
}
}
private RegisterResultEnum Register(SIPTransaction registerTransaction)
{
try
{
SIPRequest sipRequest = registerTransaction.TransactionRequest;
SIPURI registerURI = sipRequest.URI;
SIPToHeader toHeader = sipRequest.Header.To;
string toUser = toHeader.ToURI.User;
string canonicalDomain = (m_strictRealmHandling) ? GetCanonicalDomain_External(toHeader.ToURI.Host, true) : toHeader.ToURI.Host;
int requestedExpiry = GetRequestedExpiry(sipRequest);
if (canonicalDomain == null)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Warn, "Register request for " + toHeader.ToURI.Host + " rejected as no matching domain found.", null));
SIPResponse noDomainResponse = GetErrorResponse(sipRequest, SIPResponseStatusCodesEnum.Forbidden, "Domain not serviced");
registerTransaction.SendFinalResponse(noDomainResponse);
return(RegisterResultEnum.DomainNotServiced);
}
SIPAccountAsset sipAccountAsset = GetSIPAccount_External(s => s.SIPUsername == toUser && s.SIPDomain == canonicalDomain);
SIPRequestAuthenticationResult authenticationResult = SIPRequestAuthenticator_External(registerTransaction.LocalSIPEndPoint, registerTransaction.RemoteEndPoint, sipRequest, sipAccountAsset.SIPAccount, FireProxyLogEvent);
if (!authenticationResult.Authenticated)
{
// 401 Response with a fresh nonce needs to be sent.
SIPResponse authReqdResponse = SIPTransport.GetResponse(sipRequest, authenticationResult.ErrorResponse, null);
authReqdResponse.Header.AuthenticationHeader = authenticationResult.AuthenticationRequiredHeader;
registerTransaction.SendFinalResponse(authReqdResponse);
if (authenticationResult.ErrorResponse == SIPResponseStatusCodesEnum.Forbidden)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Warn, "Forbidden " + toUser + "@" + canonicalDomain + " does not exist, " + sipRequest.Header.ProxyReceivedFrom + ", " + sipRequest.Header.UserAgent + ".", null));
return(RegisterResultEnum.Forbidden);
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Registrar, "Authentication required for " + toUser + "@" + canonicalDomain + " from " + sipRequest.Header.ProxyReceivedFrom + ".", toUser));
return(RegisterResultEnum.AuthenticationRequired);
}
}
else
{
// Authenticated.
//if (!sipRequest.Header.UserAgent.IsNullOrBlank() && !m_switchboarduserAgentPrefix.IsNullOrBlank() && sipRequest.Header.UserAgent.StartsWith(m_switchboarduserAgentPrefix))
//{
// // Check that the switchboard user is authorised.
// var customer = CustomerPersistor_External.Get(x => x.CustomerUsername == sipAccount.Owner);
// if (!(customer.ServiceLevel == CustomerServiceLevels.Switchboard.ToString() || customer.ServiceLevel == CustomerServiceLevels.Gold.ToString()))
// {
// FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Warn, "Register request for switchboard from " + toHeader.ToURI.Host + " rejected as not correct service level.", sipAccount.Owner));
// SIPResponse payReqdResponse = GetErrorResponse(sipRequest, SIPResponseStatusCodesEnum.PaymentRequired, "You need to purchase a Switchboard service");
// registerTransaction.SendFinalResponse(payReqdResponse);
// return RegisterResultEnum.SwitchboardPaymentRequired;
// }
//}
if (sipRequest.Header.Contact == null || sipRequest.Header.Contact.Count == 0)
{
// No contacts header to update bindings with, return a list of the current bindings.
List <SIPRegistrarBinding> bindings = m_registrarBindingsManager.GetBindings(sipAccountAsset.Id);
//List<SIPContactHeader> contactsList = m_registrarBindingsManager.GetContactHeader(); // registration.GetContactHeader(true, null);
if (bindings != null)
{
sipRequest.Header.Contact = GetContactHeader(bindings);
}
SIPResponse okResponse = GetOkResponse(sipRequest);
registerTransaction.SendFinalResponse(okResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.RegisterSuccess, "Empty registration request successful for " + toUser + "@" + canonicalDomain + " from " + sipRequest.Header.ProxyReceivedFrom + ".", toUser));
}
else
{
SIPEndPoint uacRemoteEndPoint = SIPEndPoint.TryParse(sipRequest.Header.ProxyReceivedFrom) ?? registerTransaction.RemoteEndPoint;
SIPEndPoint proxySIPEndPoint = SIPEndPoint.TryParse(sipRequest.Header.ProxyReceivedOn);
SIPEndPoint registrarEndPoint = registerTransaction.LocalSIPEndPoint;
SIPResponseStatusCodesEnum updateResult = SIPResponseStatusCodesEnum.Ok;
string updateMessage = null;
DateTime startTime = DateTime.Now;
List <SIPRegistrarBinding> bindingsList = m_registrarBindingsManager.UpdateBindings(
sipAccountAsset.SIPAccount,
proxySIPEndPoint,
uacRemoteEndPoint,
registrarEndPoint,
//sipRequest.Header.Contact[0].ContactURI.CopyOf(),
sipRequest.Header.Contact,
sipRequest.Header.CallId,
sipRequest.Header.CSeq,
//sipRequest.Header.Contact[0].Expires,
sipRequest.Header.Expires,
sipRequest.Header.UserAgent,
out updateResult,
out updateMessage);
//int bindingExpiry = GetBindingExpiry(bindingsList, sipRequest.Header.Contact[0].ContactURI.ToString());
TimeSpan duration = DateTime.Now.Subtract(startTime);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.RegistrarTiming, "Binding update time for " + toUser + "@" + canonicalDomain + " took " + duration.TotalMilliseconds + "ms.", null));
if (updateResult == SIPResponseStatusCodesEnum.Ok)
{
string proxySocketStr = (proxySIPEndPoint != null) ? " (proxy=" + proxySIPEndPoint.ToString() + ")" : null;
int bindingCount = 1;
foreach (SIPRegistrarBinding binding in bindingsList)
{
string bindingIndex = (bindingsList.Count == 1) ? String.Empty : " (" + bindingCount + ")";
//FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.RegisterSuccess, "Registration successful for " + toUser + "@" + canonicalDomain + " from " + uacRemoteEndPoint + proxySocketStr + ", binding " + binding.ContactSIPURI.ToParameterlessString() + ";expiry=" + binding.Expiry + bindingIndex + ".", toUser));
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.RegisterSuccess, "Registration successful for " + toUser + "@" + canonicalDomain + " from " + uacRemoteEndPoint + ", binding " + binding.ContactSIPURI.ToParameterlessString() + ";expiry=" + binding.Expiry + bindingIndex + ".", toUser));
//FireProxyLogEvent(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPRegistrarBindingUpdate, toUser, uacRemoteEndPoint, sipAccount.Id.ToString()));
bindingCount++;
}
// The standard states that the Ok response should contain the list of current bindings but that breaks some UAs. As a
// compromise the list is returned with the Contact that UAC sent as the first one in the list.
bool contactListSupported = m_userAgentConfigs.GetUserAgentContactListSupport(sipRequest.Header.UserAgent);
if (contactListSupported)
{
sipRequest.Header.Contact = GetContactHeader(bindingsList);
}
else
{
// Some user agents can't match the contact header if the expiry is added to it.
sipRequest.Header.Contact[0].Expires = GetBindingExpiry(bindingsList, sipRequest.Header.Contact[0].ContactURI.ToString());;
}
SIPResponse okResponse = GetOkResponse(sipRequest);
// If a request was made for a switchboard token and a certificate is available to sign the tokens then generate it.
//if (sipRequest.Header.SwitchboardTokenRequest > 0 && m_switchbboardRSAProvider != null)
//{
// SwitchboardToken token = new SwitchboardToken(sipRequest.Header.SwitchboardTokenRequest, sipAccount.Owner, uacRemoteEndPoint.Address.ToString());
// lock (m_switchbboardRSAProvider)
// {
// token.SignedHash = Convert.ToBase64String(m_switchbboardRSAProvider.SignHash(Crypto.GetSHAHash(token.GetHashString()), null));
// }
// string tokenXML = token.ToXML(true);
// logger.Debug("Switchboard token set for " + sipAccount.Owner + " with expiry of " + token.Expiry + "s.");
// okResponse.Header.SwitchboardToken = Crypto.SymmetricEncrypt(sipAccount.SIPPassword, sipRequest.Header.AuthenticationHeader.SIPDigest.Nonce, tokenXML);
//}
registerTransaction.SendFinalResponse(okResponse);
}
else
{
// The binding update failed even though the REGISTER request was authorised. This is probably due to a
// temporary problem connecting to the bindings data store. Send Ok but set the binding expiry to the minimum so
// that the UA will try again as soon as possible.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Error, "Registration request successful but binding update failed for " + toUser + "@" + canonicalDomain + " from " + registerTransaction.RemoteEndPoint + ".", toUser));
sipRequest.Header.Contact[0].Expires = m_minimumBindingExpiry;
SIPResponse okResponse = GetOkResponse(sipRequest);
registerTransaction.SendFinalResponse(okResponse);
}
}
return(RegisterResultEnum.Authenticated);
}
}
catch (Exception excp)
{
string regErrorMessage = "Exception registrarcore registering. " + excp.Message + "\r\n" + registerTransaction.TransactionRequest.ToString();
logger.Error(regErrorMessage);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Error, regErrorMessage, null));
try
{
SIPResponse errorResponse = GetErrorResponse(registerTransaction.TransactionRequest, SIPResponseStatusCodesEnum.InternalServerError, null);
registerTransaction.SendFinalResponse(errorResponse);
}
catch { }
return(RegisterResultEnum.Error);
}
}
public void AddSubscribeRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest subscribeRequest)
{
try
{
if (subscribeRequest.Method != SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notSupportedResponse = SIPTransport.GetResponse(
subscribeRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, "Subscribe requests only");
m_sipTransport.SendResponse(notSupportedResponse);
}
else
{
#region Do as many validation checks as possible on the request before adding it to the queue.
if (subscribeRequest.Header.Event.IsNullOrBlank() ||
!(subscribeRequest.Header.Event.ToLower() == SIPEventPackage.Dialog.ToString().ToLower() ||
subscribeRequest.Header.Event.ToLower() == SIPEventPackage.Presence.ToString().ToLower()))
{
SIPResponse badEventResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.BadEvent, null);
m_sipTransport.SendResponse(badEventResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Event type " + subscribeRequest.Header.Event + " not supported for " + subscribeRequest.URI.ToString() + ".", null));
}
else if (subscribeRequest.Header.Expires > 0 && subscribeRequest.Header.Expires < MIN_SUBSCRIPTION_EXPIRY)
{
SIPResponse tooBriefResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.IntervalTooBrief, null);
tooBriefResponse.Header.MinExpires = MIN_SUBSCRIPTION_EXPIRY;
m_sipTransport.SendResponse(tooBriefResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Subscribe request was rejected as interval too brief " + subscribeRequest.Header.Expires + ".", null));
}
else if (subscribeRequest.Header.Contact == null || subscribeRequest.Header.Contact.Count == 0)
{
SIPResponse noContactResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing Contact header");
m_sipTransport.SendResponse(noContactResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Subscribe request was rejected due to no Contact header.", null));
}
#endregion
else
{
if (m_notifierQueue.Count < MAX_NOTIFIER_QUEUE_SIZE)
{
SIPNonInviteTransaction subscribeTransaction = m_sipTransport.CreateNonInviteTransaction(
subscribeRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
lock (m_notifierQueue)
{
m_notifierQueue.Enqueue(subscribeTransaction);
}
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeQueued, "Subscribe queued for " + subscribeRequest.Header.To.ToURI.ToString() + ".", null));
}
else
{
logger.Error("Subscribe queue exceeded max queue size " + MAX_NOTIFIER_QUEUE_SIZE + ", overloaded response sent.");
SIPResponse overloadedResponse = SIPTransport.GetResponse(subscribeRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, "Notifier overloaded, please try again shortly");
m_sipTransport.SendResponse(overloadedResponse);
}
m_notifierARE.Set();
}
}
}
catch (Exception excp)
{
logger.Error("Exception AddNotifierRequest (" + remoteEndPoint.ToString() + "). " + excp.Message);
}
}
public async Task HangupUserAgentUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport(false, MockSIPDNSManager.Resolve);
MockSIPChannel mockChannel = new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0));
transport.AddSIPChannel(mockChannel);
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 343" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF +
"Min-SE: 90" + m_CRLF +
"" + m_CRLF +
"v=0" + m_CRLF +
"o=- 3785527268 3785527269 IN IP4 192.168.11.50" + m_CRLF +
"s=pjmedia" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 4032 RTP/AVP 0 101" + m_CRLF +
"c=IN IP4 192.168.11.50" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
SIPServerUserAgent mockUas = new SIPServerUserAgent(transport, null, null, null, SIPCallDirection.In, null, null, null, uasTx);
await userAgent.Answer(mockUas, CreateMediaSession());
// Incremented Cseq and modified Via header from original request. Means the request is the same dialog but different tx.
string inviteReqStr2 = "BYE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3700" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3533 BYE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"";
mockChannel.FireMessageReceived(dummySipEndPoint, dummySipEndPoint, Encoding.UTF8.GetBytes(inviteReqStr2));
}
private void Subscribe(SIPTransaction subscribeTransaction)
{
try
{
SIPRequest sipRequest = subscribeTransaction.TransactionRequest;
string fromUser = sipRequest.Header.From.FromURI.User;
string fromHost = sipRequest.Header.From.FromURI.Host;
string canonicalDomain = GetCanonicalDomain_External(fromHost, true);
if (canonicalDomain == null)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Subscribe request for " + fromHost + " rejected as no matching domain found.", null));
SIPResponse noDomainResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Forbidden, "Domain not serviced");
subscribeTransaction.SendFinalResponse(noDomainResponse);
return;
}
/** AGL */
SIPAccount sipAccount = m_sipAssetPersistor.Get(s => s.SIPUsername == fromUser && s.SIPDomain == canonicalDomain);
//SIPAccount sipAccount = SipServicesSimul.Services.SipHelper.ServerAccount;
SIPRequestAuthenticationResult authenticationResult = SIPRequestAuthenticator_External(subscribeTransaction.LocalSIPEndPoint, subscribeTransaction.RemoteEndPoint, sipRequest, sipAccount, FireProxyLogEvent);
if (!authenticationResult.Authenticated)
{
// 401 Response with a fresh nonce needs to be sent.
SIPResponse authReqdResponse = SIPTransport.GetResponse(sipRequest, authenticationResult.ErrorResponse, null);
authReqdResponse.Header.AuthenticationHeader = authenticationResult.AuthenticationRequiredHeader;
subscribeTransaction.SendFinalResponse(authReqdResponse);
if (authenticationResult.ErrorResponse == SIPResponseStatusCodesEnum.Forbidden)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Forbidden " + fromUser + "@" + canonicalDomain + " does not exist, " + sipRequest.Header.ProxyReceivedFrom.ToString() + ", " + sipRequest.Header.UserAgent + ".", null));
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Authentication required for " + fromUser + "@" + canonicalDomain + " from " + subscribeTransaction.RemoteEndPoint + ".", sipAccount.Owner));
}
return;
}
else
{
if (sipRequest.Header.To.ToTag != null)
{
// Request is to renew an existing subscription.
SIPResponseStatusCodesEnum errorResponse = SIPResponseStatusCodesEnum.None;
string errorResponseReason = null;
string sessionID = m_subscriptionsManager.RenewSubscription(sipRequest, out errorResponse, out errorResponseReason);
if (errorResponse != SIPResponseStatusCodesEnum.None)
{
// A subscription renewal attempt failed
SIPResponse renewalErrorResponse = SIPTransport.GetResponse(sipRequest, errorResponse, errorResponseReason);
subscribeTransaction.SendFinalResponse(renewalErrorResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription renewal failed for event type " + sipRequest.Header.Event + " " + sipRequest.URI.ToString() + ", " + errorResponse + " " + errorResponseReason + ".", sipAccount.Owner));
}
else if (sipRequest.Header.Expires == 0)
{
// Existing subscription was closed.
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
subscribeTransaction.SendFinalResponse(okResponse);
}
else
{
// Existing subscription was found.
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
subscribeTransaction.SendFinalResponse(okResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeRenew, "Subscription renewal for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + " and expiry " + sipRequest.Header.Expires + ".", sipAccount.Owner));
m_subscriptionsManager.SendFullStateNotify(sessionID);
}
}
else
{
// Authenticated but the this is a new subscription request and authorisation to subscribe to the requested resource also needs to be checked.
SIPURI canonicalResourceURI = sipRequest.URI.CopyOf();
string resourceCanonicalDomain = GetCanonicalDomain_External(canonicalResourceURI.Host, true);
canonicalResourceURI.Host = resourceCanonicalDomain;
SIPAccount resourceSIPAccount = null;
if (resourceCanonicalDomain == null)
{
SIPResponse notFoundResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain " + resourceCanonicalDomain + " not serviced");
subscribeTransaction.SendFinalResponse(notFoundResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", domain not serviced.", sipAccount.Owner));
return;
}
if (canonicalResourceURI.User != m_wildcardUser)
{
/** AGL */
resourceSIPAccount = SipServicesSimul.Services.SipHelper.ServerAccount;
//resourceSIPAccount = m_sipAssetPersistor.Get(s => s.SIPUsername == canonicalResourceURI.User && s.SIPDomain == canonicalResourceURI.Host);
if (resourceSIPAccount == null)
{
SIPResponse notFoundResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Requested resource does not exist");
subscribeTransaction.SendFinalResponse(notFoundResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", SIP account does not exist.", sipAccount.Owner));
return;
}
}
// Check the owner permissions on the requesting and subscribed resources.
bool authorised = false;
string adminID = null;
if (canonicalResourceURI.User == m_wildcardUser || sipAccount.Owner == resourceSIPAccount.Owner)
{
authorised = true;
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Subscription to " + canonicalResourceURI.ToString() + " authorised due to common owner.", sipAccount.Owner));
}
else
{
// Lookup the customer record for the requestor and check the administrative level on it.
Customer requestingCustomer = GetCustomer_External(c => c.CustomerUsername == sipAccount.Owner);
adminID = requestingCustomer.AdminId;
if (!resourceSIPAccount.AdminMemberId.IsNullOrBlank() && requestingCustomer.AdminId == resourceSIPAccount.AdminMemberId)
{
authorised = true;
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Subscription to " + canonicalResourceURI.ToString() + " authorised due to requestor admin permissions for domain " + resourceSIPAccount.AdminMemberId + ".", sipAccount.Owner));
}
else if (requestingCustomer.AdminId == m_topLevelAdminID)
{
authorised = true;
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Subscription to " + canonicalResourceURI.ToString() + " authorised due to requestor having top level admin permissions.", sipAccount.Owner));
}
}
if (authorised)
{
// Request is to create a new subscription.
SIPResponseStatusCodesEnum errorResponse = SIPResponseStatusCodesEnum.None;
string errorResponseReason = null;
string toTag = CallProperties.CreateNewTag();
string sessionID = m_subscriptionsManager.SubscribeClient(sipAccount.Owner, adminID, sipRequest, toTag, canonicalResourceURI, out errorResponse, out errorResponseReason);
if (errorResponse != SIPResponseStatusCodesEnum.None)
{
SIPResponse subscribeErrorResponse = SIPTransport.GetResponse(sipRequest, errorResponse, errorResponseReason);
subscribeTransaction.SendFinalResponse(subscribeErrorResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAccept, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", " + errorResponse + " " + errorResponseReason + ".", sipAccount.Owner));
}
else
{
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.To.ToTag = toTag;
okResponse.Header.Expires = sipRequest.Header.Expires;
okResponse.Header.Contact = new List <SIPContactHeader>()
{
new SIPContactHeader(null, new SIPURI(SIPSchemesEnum.sip, subscribeTransaction.LocalSIPEndPoint))
};
subscribeTransaction.SendFinalResponse(okResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAccept, "Subscription accepted for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + " and expiry " + sipRequest.Header.Expires + ".", sipAccount.Owner));
if (sessionID != null)
{
m_subscriptionsManager.SendFullStateNotify(sessionID);
}
}
}
else
{
SIPResponse forbiddenResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Forbidden, "Requested resource not authorised");
subscribeTransaction.SendFinalResponse(forbiddenResponse);
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", requesting account " + sipAccount.Owner + " was not authorised.", sipAccount.Owner));
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception notifiercore subscribing. " + excp.Message + "\r\n" + subscribeTransaction.TransactionRequest.ToString());
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Error, "Exception notifiercore subscribing. " + excp.Message, null));
SIPResponse errorResponse = SIPTransport.GetResponse(subscribeTransaction.TransactionRequest, SIPResponseStatusCodesEnum.InternalServerError, null);
subscribeTransaction.SendFinalResponse(errorResponse);
}
}
public void Progress(SIPResponseStatusCodesEnum progressStatus, string reasonPhrase, string[] customHeaders, string progressContentType, string progressBody)
{
try
{
if (!IsUASAnswered)
{
if ((int)progressStatus >= 200)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "UAS call was passed an invalid response status of " + (int)progressStatus + ", ignoring.", m_owner));
}
else
{
if (UASStateChanged != null)
{
UASStateChanged(this, progressStatus, reasonPhrase);
}
// Allow all Trying responses through as some may contain additional useful information on the call state for the caller.
// Also if the response is a 183 Session Progress with audio forward it.
if (m_uasTransaction.TransactionState == SIPTransactionStatesEnum.Proceeding && progressStatus != SIPResponseStatusCodesEnum.Trying &&
!(progressStatus == SIPResponseStatusCodesEnum.SessionProgress && progressBody != null))
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "UAS call ignoring progress response with status of " + (int)progressStatus + " as already in " + m_uasTransaction.TransactionState + ".", m_owner));
}
else
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "UAS call progressing with " + progressStatus + ".", m_owner));
SIPResponse progressResponse = SIPTransport.GetResponse(m_uasTransaction.TransactionRequest, progressStatus, reasonPhrase);
if (progressResponse.Status != SIPResponseStatusCodesEnum.Trying)
{
progressResponse.Header.To.ToTag = m_uasTransaction.LocalTag;
}
if (!progressBody.IsNullOrBlank())
{
progressResponse.Body = progressBody;
progressResponse.Header.ContentType = progressContentType;
progressResponse.Header.ContentLength = progressBody.Length;
}
if (customHeaders != null && customHeaders.Length > 0)
{
foreach (string header in customHeaders)
{
progressResponse.Header.UnknownHeaders.Add(header);
}
}
m_uasTransaction.SendInformationalResponse(progressResponse);
}
}
}
else
{
logger.Warn("SIPServerUserAgent Progress fired on already answered call.");
}
}
catch (Exception excp)
{
logger.Error("Exception SIPServerUserAgent Progress. " + excp.Message);
}
}
static void Main()
{
Console.WriteLine("SIPSorcery Blind Transfer Demo: Transferee");
Console.WriteLine("Waiting for incoming call from Transferor.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(_sipTransport);
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => Log.LogDebug("Call hungup by remote party.");
userAgent.OnIncomingCall += async(ua, req) =>
{
var rtpAudioSession = new AudioSendOnlyMediaSession();
var uas = ua.AcceptCall(req);
bool answerResult = await ua.Answer(uas, rtpAudioSession);
Log.LogDebug($"Answer incoming call result {answerResult}.");
_ = Task.Run(async() =>
{
await Task.Delay(1000);
Log.LogDebug($"Sending DTMF sequence {string.Join("", DTMF_SEQUENCEFOR_TRANSFEROR.Select(x => x))}.");
foreach (byte dtmf in DTMF_SEQUENCEFOR_TRANSFEROR)
{
Log.LogDebug($"Sending DTMF tone to transferor {dtmf}.");
await ua.SendDtmf(dtmf);
}
});
};
userAgent.OnTransferRequested += (referredTo, referredBy) => true;
userAgent.OnTransferToTargetSuccessful += (dst) =>
{
Task.Run(async() =>
{
await Task.Delay(1000);
Log.LogDebug($"Sending DTMF sequence {string.Join("", DTMF_SEQUENCEFOR_TRANSFEROR.Select(x => x))}.");
foreach (byte dtmf in DTMF_SEQUENCEFOR_TRANSFEROR)
{
Log.LogDebug($"Sending DTMF tone to target {dtmf}.");
await userAgent.SendDtmf(dtmf);
}
});
};
Task.Run(() => OnKeyPress(userAgent, exitCts));
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
//userAgent?.Hangup();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
if (_sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
_sipTransport.Shutdown();
}
#endregion
}
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.HasMediaFormat((int)SDPMediaFormatsEnum.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, null, null, SIPCallDirection.In, null, null, uasTransaction);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.Close(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 = 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();
}
public void GotRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
// Used in the proxy monitor messages only, plays no part in request routing.
string fromUser = (sipRequest.Header.From != null) ? sipRequest.Header.From.FromURI.User : null;
string fromURIStr = (sipRequest.Header.From != null) ? sipRequest.Header.From.FromURI.ToString() : "null";
//string toUser = (sipRequest.Header.To != null) ? sipRequest.Header.To.ToURI.User : null;
//string summaryStr = "req " + sipRequest.Method + " from=" + fromUser + ", to=" + toUser + ", " + remoteEndPoint.ToString();
//logger.Debug("AppServerCore GotRequest " + sipRequest.Method + " from " + remoteEndPoint.ToString() + " callid=" + sipRequest.Header.CallId + ".");
SIPDialogue dialogue = null;
// Check dialogue requests for an existing dialogue.
if ((sipRequest.Method == SIPMethodsEnum.BYE || sipRequest.Method == SIPMethodsEnum.INFO || sipRequest.Method == SIPMethodsEnum.INVITE ||
sipRequest.Method == SIPMethodsEnum.MESSAGE || sipRequest.Method == SIPMethodsEnum.NOTIFY || sipRequest.Method == SIPMethodsEnum.OPTIONS ||
sipRequest.Method == SIPMethodsEnum.REFER) &&
sipRequest.Header.From != null && sipRequest.Header.From.FromTag != null && sipRequest.Header.To != null && sipRequest.Header.To.ToTag != null)
{
dialogue = m_sipDialogueManager.GetDialogue(sipRequest);
}
if (dialogue != null && sipRequest.Method != SIPMethodsEnum.ACK)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Matching dialogue found for " + sipRequest.Method + " to " + sipRequest.URI.ToString() + " from " + remoteEndPoint + ".", dialogue.Owner));
if (sipRequest.Method != SIPMethodsEnum.REFER)
{
m_sipDialogueManager.ProcessInDialogueRequest(localSIPEndPoint, remoteEndPoint, sipRequest, dialogue);
}
else
{
m_sipDialogueManager.ProcessInDialogueReferRequest(localSIPEndPoint, remoteEndPoint, sipRequest, dialogue, m_callManager.BlindTransfer);
}
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
#region CANCEL request handling.
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Cancelling call for " + sipRequest.URI.ToString() + ".", fromUser));
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".", fromUser));
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
#endregion
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No dialogue matched for BYE to " + sipRequest.URI.ToString() + ".", fromUser));
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REFER)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No dialogue matched for REFER to " + sipRequest.URI.ToString() + ".", fromUser));
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.ACK)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No transaction matched for ACK for " + sipRequest.URI.ToString() + ".", fromUser));
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
#region INVITE request processing.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "AppServerCore INVITE received, uri=" + sipRequest.URI.ToString() + ", cseq=" + sipRequest.Header.CSeq + ".", null));
if (sipRequest.URI.User == m_dispatcherUsername)
{
// Incoming call from monitoring process checking the application server is still running.
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy, true);
//uasTransaction.CDR = null;
SIPServerUserAgent incomingCall = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, sipRequest.URI.User, sipRequest.URI.Host, SIPCallDirection.In, null, null, null, uasTransaction);
//incomingCall.NoCDR();
uasTransaction.NewCallReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(incomingCall); };
uasTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else if (GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false) != null)
{
// Call identified as outgoing call for application server serviced domain.
string fromDomain = GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPServerUserAgent outgoingCall = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, fromUser, fromDomain, SIPCallDirection.Out, GetSIPAccount_External, SIPRequestAuthenticator.AuthenticateSIPRequest, FireProxyLogEvent, uasTransaction);
uasTransaction.NewCallReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(outgoingCall); };
uasTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else if (GetCanonicalDomain_External(sipRequest.URI.Host, true) != null)
{
// Call identified as incoming call for application server serviced domain.
if (sipRequest.URI.User.IsNullOrBlank())
{
// Cannot process incoming call if no user is specified.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "INVITE received with an empty URI user " + sipRequest.URI.ToString() + ", returning address incomplete.", null));
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPResponse addressIncompleteResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.AddressIncomplete, "No user specified");
uasTransaction.SendFinalResponse(addressIncompleteResponse);
}
else
{
// Send the incoming call to the call manager for processing.
string uriUser = sipRequest.URI.User;
string uriDomain = GetCanonicalDomain_External(sipRequest.URI.Host, true);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPServerUserAgent incomingCall = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, uriUser, uriDomain, SIPCallDirection.In, GetSIPAccount_External, null, FireProxyLogEvent, uasTransaction);
uasTransaction.NewCallReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(incomingCall); };
uasTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
}
else
{
// Return not found for non-serviced domain.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Domain not serviced " + sipRequest.URI.ToString() + ", returning not found.", null));
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPResponse notServicedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain not serviced");
uasTransaction.SendFinalResponse(notServicedResponse);
}
#endregion
}
else if (sipRequest.Method == SIPMethodsEnum.MESSAGE)
{
#region Processing non-INVITE requests that are accepted by the dialplan processing engine.
if (GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false) != null)
{
// Call identified as outgoing request for application server serviced domain.
string fromDomain = GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false);
SIPNonInviteTransaction nonInviteTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPNonInviteServerUserAgent outgoingRequest = new SIPNonInviteServerUserAgent(m_sipTransport, m_outboundProxy, fromUser, fromDomain, SIPCallDirection.Out, GetSIPAccount_External, SIPRequestAuthenticator.AuthenticateSIPRequest, FireProxyLogEvent, nonInviteTransaction);
nonInviteTransaction.NonInviteRequestReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(outgoingRequest); };
nonInviteTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else if (GetCanonicalDomain_External(sipRequest.URI.Host, true) != null)
{
// Call identified as incoming call for application server serviced domain.
if (sipRequest.URI.User.IsNullOrBlank())
{
// Cannot process incoming call if no user is specified.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, sipRequest.Method + " request received with an empty URI user " + sipRequest.URI.ToString() + ", returning address incomplete.", null));
SIPResponse addressIncompleteResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.AddressIncomplete, "No user specified");
m_sipTransport.SendResponse(addressIncompleteResponse);
}
else
{
// Send the incoming call to the call manager for processing.
string uriUser = sipRequest.URI.User;
string uriDomain = GetCanonicalDomain_External(sipRequest.URI.Host, true);
SIPNonInviteTransaction nonInviteTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPNonInviteServerUserAgent incomingRequest = new SIPNonInviteServerUserAgent(m_sipTransport, m_outboundProxy, uriUser, uriDomain, SIPCallDirection.In, GetSIPAccount_External, null, FireProxyLogEvent, nonInviteTransaction);
nonInviteTransaction.NonInviteRequestReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(incomingRequest); };
nonInviteTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
}
else
{
// Return not found for non-serviced domain.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Domain not serviced " + sipRequest.URI.ToString() + ", returning not found.", null));
SIPResponse notServicedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain not serviced");
m_sipTransport.SendResponse(notServicedResponse);
}
#endregion
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.UnrecognisedMessage, "MethodNotAllowed response for " + sipRequest.Method + " from " + fromUser + " socket " + remoteEndPoint.ToString() + ".", null));
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
catch (Exception excp)
{
string reqExcpError = "Exception SIPAppServerCore GotRequest (" + remoteEndPoint + "). " + excp.Message;
logger.Error(reqExcpError);
SIPMonitorEvent reqExcpEvent = new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.Error, reqExcpError, sipRequest, null, localSIPEndPoint, remoteEndPoint, SIPCallDirection.In);
FireProxyLogEvent(reqExcpEvent);
throw excp;
}
}
public SIPTransactionEngine(SIPTransport sipTransport)
{
m_sipTransport = sipTransport;
Task.Factory.StartNew(ProcessPendingTransactions, TaskCreationOptions.LongRunning);
}
public virtual void Initialize(SIPAuthenticateRequestDelegate sipRequestAuthenticator, SIPAccount account, SIPTransport trasnport)
{
}