public SIPToXMPPCall(SIPServerUserAgent uas, XMPPPhoneSession xmppCall, SIPTransport sipTransport, IPAddress ipAddress)
{
m_uas = uas;
m_xmppCall = xmppCall;
m_sipTransport = sipTransport;
m_ipAddress = ipAddress;
m_uas.CallCancelled += SIPCallCancelled;
m_xmppCall.Accepted += Answered;
m_xmppCall.Rejected += CallFailed;
m_xmppCall.Hungup += Hangup;
}
/// <summary>
/// The current call has ended. Reset the state of the user agent.
/// </summary>
private void CallEnded()
{
m_uac = null;
m_uas = null;
if (MediaSession != null && !MediaSession.IsClosed)
{
MediaSession.Close("normal");
MediaSession = null;
}
OnCallHungup?.Invoke();
}
/// <summary>
/// The current call has ended. Reset the state of the user agent.
/// </summary>
private void CallEnded()
{
m_uac = null;
m_uas = null;
if (MediaSession != null)
{
MediaSession.SessionMediaChanged -= MediaSessionOnSessionMediaChanged;
MediaSession.Close();
MediaSession = null;
}
OnCallHungup?.Invoke();
}
/// <summary>
/// This method can be used to start the processing of a new incoming call request.
/// The user agent will is acting as a server for this operation and it can be considered
/// the opposite of the Call method. This is only the first step in answering an incoming
/// call. It can still be rejected or answered after this point.
/// </summary>
/// <param name="inviteRequest">The invite request representing the incoming call.</param>
/// <returns>An ID string that needs to be supplied when the call is answered or rejected
/// (used to manage multiple pending incoming calls).</returns>
public SIPServerUserAgent AcceptCall(SIPRequest inviteRequest)
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(m_transport, inviteRequest, m_outboundProxy);
SIPServerUserAgent uas = new SIPServerUserAgent(m_transport, m_outboundProxy, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.CallCancelled += (pendingUas) =>
{
CallEnded();
ServerCallCancelled?.Invoke(pendingUas);
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
return(uas);
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
/// <param name="customHeaders">Custom SIP-Headers to use in Answer.</param>
public async Task Answer(SIPServerUserAgent uas, IMediaSession mediaSession, string[] customHeaders)
{
// This call is now taking over any existing call.
if (IsCallActive)
{
Hangup();
}
else if (uas.IsCancelled)
{
logger.LogDebug("The incoming call has been cancelled.");
mediaSession?.Close("call cancelled");
}
else
{
m_cts = new CancellationTokenSource();
var sipRequest = uas.ClientTransaction.TransactionRequest;
MediaSession = mediaSession;
MediaSession.OnRtpEvent += OnRemoteRtpEvent;
//MediaSession.OnRtpClosed += (reason) => Hangup();
MediaSession.OnRtpClosed += (reason) =>
{
if (!MediaSession.IsClosed)
{
logger.LogWarning($"RTP channel was closed with reason {reason}.");
}
};
SDP remoteSdp = SDP.ParseSDPDescription(sipRequest.Body);
MediaSession.setRemoteDescription(new RTCSessionDescription {
sdp = remoteSdp, type = RTCSdpType.offer
});;
var sdpAnswer = await MediaSession.createAnswer(null).ConfigureAwait(false);
MediaSession.setLocalDescription(new RTCSessionDescription {
sdp = sdpAnswer, type = RTCSdpType.answer
});
await MediaSession.Start().ConfigureAwait(false);
m_uas = uas;
m_uas.Answer(m_sdpContentType, sdpAnswer.ToString(), null, SIPDialogueTransferModesEnum.Default, customHeaders);
Dialogue.DialogueState = SIPDialogueStateEnum.Confirmed;
}
}
/// <summary>
/// This method can be used to start the processing of a new incoming call request.
/// The user agent will is acting as a server for this operation and it can be considered
/// the opposite of the Call method.
/// </summary>
/// <param name="uasInviteTx">The invite transaction representing the incoming call.</param>
/// <returns>True if the call is accepted, false otherwise.</returns>
public bool AcceptCall(UASInviteTransaction uasInviteTx)
{
SIPServerUserAgent uas = new SIPServerUserAgent(m_transport, m_outboundProxy, null, null, SIPCallDirection.In, null, null, null, uasInviteTx);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// TODO: Decide how to deal with multiple simultaneous calls.
if (m_uas != null)
{
uas.Reject(SIPResponseStatusCodesEnum.BusyHere, null, null);
return(false);
}
else
{
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
m_uas = uas;
return(true);
}
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
public async Task Answer(SIPServerUserAgent uas, IMediaSession mediaSession)
{
// This call is now taking over any existing call.
if (IsCallActive)
{
Hangup();
}
var sipRequest = uas.ClientTransaction.TransactionRequest;
MediaSession = mediaSession;
MediaSession.SessionMediaChanged += MediaSessionOnSessionMediaChanged;
MediaSession.OnRtpClosed += (reason) => Hangup();
var sdpAnswer = await MediaSession.AnswerOffer(sipRequest.Body).ConfigureAwait(false);
m_uas = uas;
m_uas.Answer(m_sdpContentType, sdpAnswer, null, SIPDialogueTransferModesEnum.Default);
Dialogue.DialogueState = SIPDialogueStateEnum.Confirmed;
}
/// <param name="rtpListenAddress">The local IP address to establish the RTP listener socket on.</param>
/// <param name="sdpAdvertiseAddress">The public IP address to put into the SDP sent back to the caller.</param>
/// <param name="request">The INVITE request that instigated the RTP diagnostics job.</param>
public RTPDiagnosticsJob(IPAddress rtpListenAddress, IPAddress sdpAdvertiseAddress, SIPServerUserAgent uas, SIPRequest request)
{
m_request = request;
m_remoteSDP = SDP.ParseSDPDescription(request.Body);
RemoteRTPEndPoint = new IPEndPoint(IPAddress.Parse(m_remoteSDP.Connection.ConnectionAddress), m_remoteSDP.Media[0].Port);
UAS = uas;
//m_rawSourceStream = new RawSourceWaveStream(m_outStream, WaveFormat.CreateMuLawFormat(8000, 1));
//m_waveFileWriter = new WaveFileWriter("out.wav", new WaveFormat(8000, 16, 1));
m_waveFileWriter = new WaveFileWriter("out.wav", new WaveFormat(8000, 16, 1));
//m_outPCMStream = WaveFormatConversionStream.CreatePcmStream(m_rawSourceStream);
//m_rawRTPPayloadWriter = new StreamWriter("out.rtp");
//m_rawRTPPayloadReader = new StreamReader("in.rtp");
//IPEndPoint rtpListenEndPoint = null;
IPEndPoint rtpListenEndPoint = null;
NetServices.CreateRandomUDPListener(rtpListenAddress, RTP_PORTRANGE_START, RTP_PORTRANGE_END, m_inUsePorts, out rtpListenEndPoint);
RTPListenEndPoint = rtpListenEndPoint;
m_inUsePorts.Add(rtpListenEndPoint.Port);
//RTPListenEndPoint = new IPEndPoint(rtpListenAddress, RTP_PORTRANGE_START);
m_rtpChannel = new RTPChannel(RTPListenEndPoint);
m_rtpChannel.SampleReceived += SampleReceived;
ThreadPool.QueueUserWorkItem(delegate { GetAudioSamples(); });
LocalSDP = new SDP()
{
SessionId = Crypto.GetRandomString(6),
Address = sdpAdvertiseAddress.ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(sdpAdvertiseAddress.ToString()),
Media = new List<SDPMediaAnnouncement>()
{
new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
Port = RTPListenEndPoint.Port,
MediaFormats = new List<SDPMediaFormat>() { new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU) }
}
}
};
}
/// <summary>
/// Processes the callback action that is initiated by the callmanager service. The callback method is typically initiated from
/// a link on an authenticated web page and requires the user to be authenticated.
/// </summary>
/// <param name="username">The authenticated username of the user making the callback request.</param>
/// <param name="dialString1">The first leg dial string of the callback.</param>
/// <param name="dialString2">The second leg dial string of the callback.</param>
/// <returns>A string that is returned to the user making the callmanager request that inidcates what action was taken.</returns>
public string ProcessCallback(string username, string dialString1, string dialString2)
{
Customer customer = null;
try
{
customer = m_customerPersistor.Get(c => c.CustomerUsername == username);
if (customer == null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Callback rejected for " + username + " as no matching user.", null));
return "Sorry no matching user was found, the callback was not initiated.";
}
else if (customer.Suspended)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Callback rejected for " + username + " as user account is suspended.", null));
return "Sorry your account is suspended.";
}
else if (customer.ServiceLevel == CustomerServiceLevels.PremiumPayReqd.ToString() || customer.ServiceLevel == CustomerServiceLevels.ProfessionalPayReqd.ToString() || customer.ServiceLevel == CustomerServiceLevels.SwitchboardPayReqd.ToString())
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Callback rejected for " + username + ", user account requires payment.", null));
return "Sorry your account requires payment.";
}
else
{
UASInviteTransaction dummyTransaction = GetDummyWebCallbackTransaction("callback");
ISIPServerUserAgent uas = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, username, SIPDomainManager.DEFAULT_LOCAL_DOMAIN, SIPCallDirection.Out, GetSIPAccount_External, null, Log_External, dummyTransaction);
string callbackScript =
"sys.Log(\"Callback dialString1=" + dialString1 + ", dialString2=" + dialString2 + ".\")\n" +
"sys.Callback(\"" + dialString1 + "\",\"" + dialString2 + "\", 0)\n";
SIPDialPlan callbackDialPlan = new SIPDialPlan(username, null, null, callbackScript, SIPDialPlanScriptTypesEnum.Ruby);
callbackDialPlan.Id = Guid.Empty; // Prevents the increment and decrement on the execution counts.
DialPlanScriptContext scriptContext = new DialPlanScriptContext(
Log_External,
m_sipTransport,
CreateDialogueBridge,
m_outboundProxy,
uas,
callbackDialPlan,
GetSIPProviders_External(p => p.Owner == username, null, 0, Int32.MaxValue),
null,
null,
customer,
null,
GetCanonicalDomain_External);
m_dialPlanEngine.Execute(scriptContext, uas, SIPCallDirection.Out, CreateDialogueBridge, this);
return null;
}
}
catch (Exception excp)
{
logger.Error("Exception SIPCallManager ProcessCallback. " + excp.Message);
return "Sorry there was an unexpected error, the callback was not initiated.";
}
}
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.
AppendTraceMessage("Call hungup by server: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
AppendSIPTraceMessage("Request Received " + localSIPEndPoint + "<-" + remoteEndPoint + "\n" + sipRequest.ToString());
m_uac.SIPDialogue.Hangup(m_sipTransport, null);
ResetToCallStartState();
}
else if (m_uas != null && m_uas.SIPDialogue != null && sipRequest.Header.CallId == m_uas.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
AppendTraceMessage("Call hungup by client: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
AppendSIPTraceMessage("Request Received " + localSIPEndPoint + "<-" + remoteEndPoint + "\n" + sipRequest.ToString());
m_uas.SIPDialogue.Hangup(m_sipTransport, null);
ResetToCallStartState();
}
else
{
AppendTraceMessage("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)
{
ResetToCallStartState();
ClearTraces();
AppendTraceMessage("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
SetVisibility(m_uacGrid, Visibility.Collapsed);
SetVisibility(m_uasGrid, Visibility.Visible);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
m_uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, LogTraceMessage, uasTransaction);
m_uas.CallCancelled += UASCallCancelled;
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
AppendTraceMessage("Matching CANCEL request received " + sipRequest.URI.ToString() + ".\n");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
AppendTraceMessage("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
{
AppendTraceMessage("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.\n");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
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);
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;
}
}
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);
}
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
/// <param name="customHeaders">Custom SIP-Headers to use in Answer.</param>
/// <returns>True if the call was successfully answered or false if there was a problem
/// such as incompatible codecs.</returns>
public async Task <bool> Answer(SIPServerUserAgent uas, IMediaSession mediaSession, string[] customHeaders)
{
// This call is now taking over any existing call.
if (IsCallActive)
{
Hangup();
}
if (uas.IsCancelled)
{
logger.LogDebug("The incoming call has been cancelled.");
mediaSession?.Close("call cancelled");
return(false);
}
else
{
m_cts = new CancellationTokenSource();
var sipRequest = uas.ClientTransaction.TransactionRequest;
MediaSession = mediaSession;
MediaSession.OnRtpEvent += OnRemoteRtpEvent;
MediaSession.OnRtpClosed += (reason) =>
{
if (!MediaSession.IsClosed)
{
logger.LogWarning($"RTP channel was closed with reason {reason}.");
}
};
string sdp = null;
if (!String.IsNullOrEmpty(sipRequest.Body))
{
// The SDP offer was included in the INVITE request.
SDP remoteSdp = SDP.ParseSDPDescription(sipRequest.Body);
var setRemoteResult = MediaSession.SetRemoteDescription(remoteSdp);
if (setRemoteResult != SetDescriptionResultEnum.OK)
{
logger.LogWarning($"Error setting remote description from INVITE {setRemoteResult}.");
uas.Reject(SIPResponseStatusCodesEnum.NotAcceptable, setRemoteResult.ToString());
MediaSession.Close("sdp offer not acceptable");
Hangup();
return(false);
}
else
{
var sdpAnswer = MediaSession.CreateAnswer(null);
sdp = sdpAnswer.ToString();
}
}
else
{
// No SDP offer was included in the INVITE request need to wait for the ACK.
var sdpAnnounceAddress = NetServices.GetLocalAddressForRemote(sipRequest.RemoteSIPEndPoint.GetIPEndPoint().Address);
var sdpOffer = MediaSession.CreateOffer(sdpAnnounceAddress);
sdp = sdpOffer.ToString();
}
m_uas = uas;
// In cases where the initial INVITE did not contain an SDP offer the action sequence is:
// - INVITE with no SDP offer received,
// - Reply with OK and an SDP offer,
// - Wait for ACK with SDP answer.
TaskCompletionSource <SIPDialogue> dialogueCreatedTcs = new TaskCompletionSource <SIPDialogue>(TaskCreationOptions.RunContinuationsAsynchronously);
m_uas.OnDialogueCreated += (dialogue) => dialogueCreatedTcs.TrySetResult(dialogue);
m_uas.Answer(m_sdpContentType, sdp, null, SIPDialogueTransferModesEnum.Default, customHeaders);
await Task.WhenAny(dialogueCreatedTcs.Task, Task.Delay(WAIT_DIALOG_TIMEOUT)).ConfigureAwait(false);
if (Dialogue != null)
{
if (MediaSession.RemoteDescription == null)
{
// If the initial INVITE did not contain an offer then the remote description will not yet be set.
var remoteSDP = SDP.ParseSDPDescription(Dialogue.RemoteSDP);
var setRemoteResult = MediaSession.SetRemoteDescription(remoteSDP);
if (setRemoteResult != SetDescriptionResultEnum.OK)
{
// Failed to set the remote SDP from the ACK request. Only option is to hangup.
logger.LogWarning($"Error setting remote description from ACK {setRemoteResult}.");
MediaSession.Close(setRemoteResult.ToString());
Hangup();
return(false);
}
else
{
// SDP from the ACK request was accepted. Start the RTP session.
Dialogue.DialogueState = SIPDialogueStateEnum.Confirmed;
await MediaSession.Start().ConfigureAwait(false);
return(true);
}
}
else
{
Dialogue.DialogueState = SIPDialogueStateEnum.Confirmed;
await MediaSession.Start().ConfigureAwait(false);
return(true);
}
}
else
{
logger.LogWarning("The attempt to answer a call failed as the dialog was not created. The likely cause is the ACK not being received in time.");
MediaSession.Close("dialog creation failed");
Hangup();
return(false);
}
}
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
public Task <bool> Answer(SIPServerUserAgent uas, IMediaSession mediaSession)
{
return(Answer(uas, mediaSession, null));
}
/// <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);
}
}
/// <summary>
/// Cleans up after a SIP call has completely finished.
/// </summary>
private void CallFinished()
{
if (_mediaManager != null)
{
_mediaManager.EndCall();
_mediaManager = null;
}
m_uac = null;
m_uas = null;
CallEnded();
}
/// <summary>
/// Handler for when an in dialog request is received on an established call.
/// Typical types of request will be re-INVITES for things like putting a call on or
/// off hold and REFER requests for transfers. Some in dialog request types, such
/// as re-INVITES have specific events so they can be bubbled up to the
/// application to deal with.
/// </summary>
/// <param name="request">The in dialog request received.</param>
public async Task InDialogRequestReceivedAsync(SIPRequest sipRequest)
{
// Make sure the request matches our dialog and is not a stray.
// A dialog request should match on to tag, from tag and call ID. We'll be more
// accepting just in case the sender got the tags wrong.
if (Dialogue == null || sipRequest.Header.CallId != Dialogue.CallId)
{
var noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
var sendResult = await SendResponse(noCallLegResponse);
if (sendResult != SocketError.Success)
{
logger.LogWarning($"SIPUserAgent send response failed in InCallRequestReceivedAsync with {sendResult}.");
}
}
else
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
logger.LogDebug($"Matching dialogue found for {sipRequest.StatusLine}.");
SIPNonInviteTransaction byeTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallHungup?.Invoke();
m_uac = null;
m_uas = null;
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
logger.LogDebug($"Re-INVITE request received {sipRequest.StatusLine}.");
UASInviteTransaction reInviteTransaction = m_transport.CreateUASTransaction(sipRequest, m_outboundProxy);
if (OnReinviteRequest == null)
{
// The application isn't prepared to accept re-INVITE requests. We'll reject as gently as we can to try and not lose the call.
SIPResponse notAcceptableResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptable, null);
reInviteTransaction.SendFinalResponse(notAcceptableResponse);
}
else
{
SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
reInviteTransaction.SendProvisionalResponse(tryingResponse);
OnReinviteRequest(reInviteTransaction);
}
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "OPTIONS request for established dialogue " + dialogue.DialogueName + ".", dialogue.Owner));
SIPNonInviteTransaction optionsTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Body = Dialogue.RemoteSDP;
okResponse.Header.ContentLength = okResponse.Body.Length;
okResponse.Header.ContentType = m_sdpContentType;
optionsTransaction.SendFinalResponse(okResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.MESSAGE)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "MESSAGE for call " + sipRequest.URI.ToString() + ": " + sipRequest.Body + ".", dialogue.Owner));
SIPNonInviteTransaction messageTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
messageTransaction.SendFinalResponse(okResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REFER)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "REFER received on dialogue " + dialogue.DialogueName + ", transfer mode is " + dialogue.TransferMode + ".", dialogue.Owner));
SIPNonInviteTransaction referTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
if (sipRequest.Header.ReferTo.IsNullOrBlank())
{
// A REFER request must have a Refer-To header.
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Bad REFER request, no Refer-To header.", dialogue.Owner));
SIPResponse invalidResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing mandatory Refer-To header");
referTransaction.SendFinalResponse(invalidResponse);
}
else
{
//TODO: Add handling logic for indialog REFER requests.
}
}
}
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
public async Task Answer(SIPServerUserAgent uas, IMediaSession mediaSession)
{
await Answer(uas, mediaSession, null).ConfigureAwait(false);
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
public async Task Answer(SIPServerUserAgent uas, IMediaSession mediaSession)
{
await Answer(uas, mediaSession, null);
}
private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Console.WriteLine("INVITE received from " + localSIPEndPoint.ToString());
IPEndPoint sipPhoneEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPServerUserAgent uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
uasTransaction.SendInformationalResponse(tryingResponse);
if (m_xmppClient == null)
{
m_xmppClient = new XMPPClient(XMPP_SERVER, XMPP_SERVER_PORT, XMPP_REALM, m_xmppUsername, m_xmppPassword);
m_xmppClient.Disconnected += XMPPDisconnected;
m_xmppClient.IsBound += () => { XMPPPlaceCall(uas); };
ThreadPool.QueueUserWorkItem(delegate { m_xmppClient.Connect(); });
}
else
{
XMPPPlaceCall(uas);
}
}
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() + ".");
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() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Console.WriteLine("BYE request received.");
if (m_activeCalls.ContainsKey(sipRequest.Header.CallId))
{
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
m_activeCalls[sipRequest.Header.CallId].TerminateXMPPCall();
m_activeCalls.Remove(sipRequest.Header.CallId);
}
else
{
SIPResponse doesntExistResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(doesntExistResponse);
}
}
}
private static void XMPPPlaceCall(SIPServerUserAgent uas)
{
if (!uas.IsCancelled)
{
XMPPPhoneSession phoneSession = m_xmppClient.GetPhoneSession();
SIPToXMPPCall call = new SIPToXMPPCall(uas, phoneSession, m_sipTransport, m_ipAddress);
m_activeCalls.Add(uas.CallRequest.Header.CallId, call);
call.Call(uas.CallRequest.URI.User);
}
}
/// <summary>
/// Answers the call request contained in the user agent server parameter. Note the
/// <see cref="AcceptCall(SIPRequest)"/> method should be used to create the user agent server.
/// Any existing call will be hungup.
/// </summary>
/// <param name="uas">The user agent server holding the pending call to answer.</param>
/// <param name="mediaSession">The media session used for this call</param>
/// <param name="customHeaders">Custom SIP-Headers to use in Answer.</param>
public async Task Answer(SIPServerUserAgent uas, IMediaSession mediaSession, string[] customHeaders)
{
// This call is now taking over any existing call.
if (IsCallActive)
{
Hangup();
}
else if (uas.IsCancelled)
{
logger.LogDebug("The incoming call has been cancelled.");
mediaSession?.Close("call cancelled");
}
else
{
m_cts = new CancellationTokenSource();
var sipRequest = uas.ClientTransaction.TransactionRequest;
MediaSession = mediaSession;
MediaSession.OnRtpEvent += OnRemoteRtpEvent;
//MediaSession.OnRtpClosed += (reason) => Hangup();
MediaSession.OnRtpClosed += (reason) =>
{
if (!MediaSession.IsClosed)
{
logger.LogWarning($"RTP channel was closed with reason {reason}.");
}
};
string sdp = null;
if (!String.IsNullOrEmpty(sipRequest.Body))
{
// The SDP offer was included in the INVITE request.
SDP remoteSdp = SDP.ParseSDPDescription(sipRequest.Body);
MediaSession.setRemoteDescription(new RTCSessionDescription {
sdp = remoteSdp, type = RTCSdpType.offer
});
var sdpAnswer = await MediaSession.createAnswer(null).ConfigureAwait(false);
if (sdpAnswer == null)
{
logger.LogWarning($"Could not generate an SDP answer.");
m_uas.Reject(SIPResponseStatusCodesEnum.NotAcceptable, null);
return;
}
else
{
MediaSession.setLocalDescription(new RTCSessionDescription {
sdp = sdpAnswer, type = RTCSdpType.answer
});
sdp = sdpAnswer.ToString();
}
}
else
{
// No SDP offer was included in the INVITE request need to wait for the ACK.
RTCOfferOptions offerOptions = new RTCOfferOptions {
RemoteSignallingAddress = sipRequest.RemoteSIPEndPoint.GetIPEndPoint().Address
};
var sdpOffer = await MediaSession.createOffer(offerOptions).ConfigureAwait(false);
if (sdpOffer == null)
{
// This shouldn't occur unless we're unable to create an audio/video track.
logger.LogWarning($"Could not generate an SDP answer.");
m_uas.Reject(SIPResponseStatusCodesEnum.NotAcceptable, null);
return;
}
else
{
MediaSession.setLocalDescription(new RTCSessionDescription {
sdp = sdpOffer, type = RTCSdpType.offer
});
sdp = sdpOffer.ToString();
}
}
m_uas = uas;
// In cases where the initial INVITE did not contain an SDP offer the action sequence is:
// - INVITE with no SDP offer received,
// - Reply with OK and an SDP offer,
// - Wait for ACK with SDP answer.
TaskCompletionSource <SIPDialogue> dialogueCreatedTcs = new TaskCompletionSource <SIPDialogue>(TaskCreationOptions.RunContinuationsAsynchronously);
m_uas.OnDialogueCreated += (dialogue) => dialogueCreatedTcs.TrySetResult(dialogue);
m_uas.Answer(m_sdpContentType, sdp, null, SIPDialogueTransferModesEnum.Default, customHeaders);
await Task.WhenAny(dialogueCreatedTcs.Task, Task.Delay(WAIT_DIALOG_TIMEOUT)).ConfigureAwait(false);
if (Dialogue != null)
{
if (MediaSession.remoteDescription == null || MediaSession.remoteDescription.sdp == null)
{
// If the initial INVITE did not contain an offer then the remote description will not yet be set.
var remoteSDP = SDP.ParseSDPDescription(Dialogue.RemoteSDP);
MediaSession.setRemoteDescription(new RTCSessionDescription {
sdp = remoteSDP, type = RTCSdpType.answer
});
}
Dialogue.DialogueState = SIPDialogueStateEnum.Confirmed;
await MediaSession.Start().ConfigureAwait(false);
}
else
{
logger.LogWarning("The attempt to answer a call failed as the dialog was not created. The likely cause is the ACK not being received in time.");
MediaSession.Close("dialog creation failed");
Hangup();
}
}
}
/// <summary>
/// Processes actions initiated by the callmanager web service EXCEPT for the callback method.
/// </summary>
/// <param name="username">The UNAUTHENTICATED username that was specified in the callmanager request URL.</param>
/// <param name="number">The number parameter that was specified in the callmanager request URL.</param>
/// <param name="dialplanName">The dialplan to use to process web calls, typically this will be ahrd coded to a known dialplan name.</param>
/// <param name="replacesCallID">The replacesCallID parameter that was specified in the callmanager request URL.</param>
/// <returns>A string that is returned to the user making the callmanager request that inidcates what action was taken.</returns>
public string ProcessWebCall(string username, string number, string dialplanName, string replacesCallID)
{
//bool wasExecutionCountIncremented = false;
Customer customer = null;
SIPDialPlan dialPlan = null;
try
{
customer = m_customerPersistor.Get(c => c.CustomerUsername == username);
SIPDialogue replacesDialogue = (!replacesCallID.IsNullOrBlank() && customer != null) ? m_sipDialogueManager.GetDialogueRelaxed(customer.CustomerUsername, replacesCallID) : null;
if (customer == null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Web " + dialplanName + " rejected for " + username + " and " + number + ", as no matching user.", null));
return "Sorry no matching user was found, the " + dialplanName + " was not initiated.";
}
else if (customer.Suspended)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Web " + dialplanName + " rejected for " + username + " and " + number + ", user account is suspended.", null));
return "Sorry the user's account is suspended.";
}
else if (customer.ServiceLevel == CustomerServiceLevels.PremiumPayReqd.ToString() || customer.ServiceLevel == CustomerServiceLevels.ProfessionalPayReqd.ToString() || customer.ServiceLevel == CustomerServiceLevels.SwitchboardPayReqd.ToString())
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Web " + dialplanName + " rejected for " + username + " and " + number + ", user account requires payment.", null));
return "Sorry the user's account requires payment.";
}
else if (!replacesCallID.IsNullOrBlank() && replacesDialogue == null)
{
return "Sorry the blind transfer could not be initiated, the Call-ID to transfer could not be found.";
}
else
{
dialPlan = GetDialPlan_External(d => d.Owner == username && d.DialPlanName == dialplanName);
if (dialPlan == null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Web " + dialplanName + " rejected as no " + dialplanName + " dialplan exists.", username));
return "Sorry the specified user has not enabled callbacks, the callback was not initiated.";
}
else
{
if (!IsDialPlanExecutionAllowed(dialPlan, customer))
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Execution of web call for dialplan " + dialplanName + " was not processed as maximum execution count has been reached.", username));
return "Sorry the callback was not initiated, dial plan execution exceeded maximum allowed";
}
else
{
//IncrementDialPlanExecutionCount(dialPlan, customer, originalExecutionCount + 1);
//IncrementCustomerExecutionCount(customer);
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Web call for " + dialplanName + " initialising to " + number + ".", username));
ISIPServerUserAgent uas = null;
if (replacesCallID.IsNullOrBlank())
{
UASInviteTransaction dummyTransaction = GetDummyWebCallbackTransaction(number);
uas = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, username, SIPDomainManager.DEFAULT_LOCAL_DOMAIN, SIPCallDirection.Out, GetSIPAccount_External, null, Log_External, dummyTransaction);
}
else
{
SIPDialogue oppositeDialogue = m_sipDialogueManager.GetOppositeDialogue(replacesDialogue);
uas = new SIPTransferServerUserAgent(Log_External, m_sipDialogueManager.DialogueTransfer, m_sipTransport, m_outboundProxy, replacesDialogue, oppositeDialogue, number, customer.CustomerUsername, customer.AdminId);
m_inProgressTransfers.Add(oppositeDialogue, uas as SIPTransferServerUserAgent);
}
dialPlan.AuthorisedApps = customer.AuthorisedApps + ";" + dialPlan.AuthorisedApps;
DialPlanScriptContext scriptContext = new DialPlanScriptContext(
Log_External,
m_sipTransport,
CreateDialogueBridge,
m_outboundProxy,
uas,
dialPlan,
GetSIPProviders_External(p => p.Owner == username, null, 0, Int32.MaxValue),
m_traceDirectory,
null,
customer,
null,
GetCanonicalDomain_External);
//scriptContext.DialPlanComplete += () => { DecrementCustomerExecutionCount(customer); };
m_dialPlanEngine.Execute(scriptContext, uas, SIPCallDirection.Out, CreateDialogueBridge, this);
if (replacesCallID.IsNullOrBlank())
{
return "Web call was successfully initiated.";
}
else
{
return "Blind transfer was successfully initiated.";
}
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception SIPCallManager ProcessWebCall. " + excp.Message);
//if (wasExecutionCountIncremented)
//{
//DecrementDialPlanExecutionCount(dialPlan, customer.Id, originalExecutionCount);
//DecrementCustomerExecutionCount(customer);
//}
return "Sorry there was an unexpected error, the callback was not initiated.";
}
}
