public SIPDialogue Answer(string contentType, string body, SIPDialogue answeredDialogue, SIPDialogueTransferModesEnum transferMode)
{
try
{
logger.Debug("SIPB2BUserAgent Answer.");
m_sipDialogue = answeredDialogue;
UASStateChanged?.Invoke(this, SIPResponseStatusCodesEnum.Ok, null);
SIPResponse uasOkResponse = SIPTransport.GetResponse(m_uasTransaction.TransactionRequest, SIPResponseStatusCodesEnum.Ok, null);
m_uasTransaction.SendFinalResponse(uasOkResponse);
m_uasTransaction.ACKReceived(m_blackhole, m_blackhole, null);
SIPResponse uacOkResponse = SIPTransport.GetResponse(m_uacTransaction.TransactionRequest, SIPResponseStatusCodesEnum.Ok, null);
uacOkResponse.Header.Contact = new List <SIPContactHeader>()
{
new SIPContactHeader(null, new SIPURI(SIPSchemesEnum.sip, m_blackhole))
};
m_uacTransaction.GotResponse(m_blackhole, m_blackhole, uacOkResponse);
uacOkResponse.Header.ContentType = contentType;
if (!body.IsNullOrBlank())
{
uacOkResponse.Body = body;
uacOkResponse.Header.ContentLength = body.Length;
}
CallAnswered.Invoke(this, uacOkResponse);
return(null);
}
catch (Exception excp)
{
logger.Error("Exception SIPB2BUSerAgent Answer. " + excp.Message);
throw;
}
}
public bool AuthenticateCall()
{
m_isAuthenticated = false;
try
{
if (m_sipAccount == null)
{
logger.LogWarning($"Rejecting authentication required call for {m_uasTransaction.TransactionRequestFrom}, SIP account not found.");
Reject(SIPResponseStatusCodesEnum.Forbidden, null, null);
}
else
{
SIPRequest sipRequest = m_uasTransaction.TransactionRequest;
SIPEndPoint localSIPEndPoint = (!sipRequest.Header.ProxyReceivedOn.IsNullOrBlank()) ? SIPEndPoint.ParseSIPEndPoint(sipRequest.Header.ProxyReceivedOn) : sipRequest.LocalSIPEndPoint;
SIPEndPoint remoteEndPoint = (!sipRequest.Header.ProxyReceivedFrom.IsNullOrBlank()) ? SIPEndPoint.ParseSIPEndPoint(sipRequest.Header.ProxyReceivedFrom) : sipRequest.RemoteSIPEndPoint;
var authenticationResult = SIPRequestAuthenticator.AuthenticateSIPRequest(localSIPEndPoint, remoteEndPoint, sipRequest, m_sipAccount);
if (authenticationResult.Authenticated)
{
if (authenticationResult.WasAuthenticatedByIP)
{
logger.LogDebug($"New call from {remoteEndPoint} successfully authenticated by IP address.");
}
else
{
logger.LogDebug($"New call from {remoteEndPoint} successfully authenticated by digest.");
}
m_isAuthenticated = true;
}
else
{
if (sipRequest.Header.AuthenticationHeader != null)
{
logger.LogWarning($"Call not authenticated for {m_sipAccount.SIPUsername}@{m_sipAccount.SIPDomain}, responding with {authenticationResult.ErrorResponse}.");
}
// Send authorisation failure or required response
SIPResponse authReqdResponse = SIPResponse.GetResponse(sipRequest, authenticationResult.ErrorResponse, null);
authReqdResponse.Header.AuthenticationHeader = authenticationResult.AuthenticationRequiredHeader;
m_uasTransaction.SendFinalResponse(authReqdResponse);
}
}
}
catch (Exception excp)
{
logger.LogError("Exception SIPServerUserAgent AuthenticateCall. " + excp.Message);
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
return(m_isAuthenticated);
}
/// <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 Task SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest 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)
{
bool?callAccepted = IncomingCall?.Invoke(sipRequest);
if (callAccepted == false)
{
// All user agents were 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
{
logger.Debug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
return(SIPTransport.SendResponseAsync(notAllowedResponse));
}
return(Task.FromResult(0));
}
/// <summary>
/// Event handler for receiving a re-INVITE request on an established call.
/// In call requests can be used for multitude of different purposes. In this
/// example program we're only concerned with re-INVITE requests being used
/// to place a call on/off hold.
/// </summary>
/// <param name="uasTransaction">The user agent server invite transaction that
/// was created for the request. It needs to be used for sending responses
/// to ensure reliable delivery.</param>
private static void ReinviteRequestReceived(UASInviteTransaction uasTransaction)
{
SIPRequest reinviteRequest = uasTransaction.TransactionRequest;
// Re-INVITEs can also be changing the RTP end point. We can update this each time.
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(reinviteRequest.Body);
_remoteRtpEndPoint = dstRtpEndPoint;
// If the RTP callfow attribute has changed it's most likely due to being placed on/off hold.
SDP newSDP = SDP.ParseSDPDescription(reinviteRequest.Body);
if (GetRTPStatusAttribute(newSDP) == RTP_ATTRIBUTE_SENDONLY)
{
Log.LogInformation("Remote call party has placed us on hold.");
_holdStatus = HoldStatus.RemotePutOnHold;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_RECVONLY);
var okResponse = SIPTransport.GetResponse(reinviteRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.ContentType = SDP.SDP_MIME_CONTENTTYPE;
okResponse.Body = _ourSDP.ToString();
uasTransaction.SendFinalResponse(okResponse);
}
else if (GetRTPStatusAttribute(newSDP) == RTP_ATTRIBUTE_SENDRECV && _holdStatus != HoldStatus.None)
{
Log.LogInformation("Remote call party has taken us off hold.");
_holdStatus = HoldStatus.None;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
var okResponse = SIPTransport.GetResponse(reinviteRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.ContentType = SDP.SDP_MIME_CONTENTTYPE;
okResponse.Body = _ourSDP.ToString();
uasTransaction.SendFinalResponse(okResponse);
}
else
{
Log.LogWarning("Not sure what the remote call party wants us to do...");
// We'll just reply Ok and hope eveything is good.
var okResponse = SIPTransport.GetResponse(reinviteRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.ContentType = SDP.SDP_MIME_CONTENTTYPE;
okResponse.Body = _ourSDP.ToString();
uasTransaction.SendFinalResponse(okResponse);
}
}
/// <summary>
/// Attempts to lookup the caller based on the "To" header host. If the host matches a
/// hosted domain then an attempt will be made to retrieve the SIP account for the caller.
/// If a SIP account is not found for a hosted domain the call is rejected with a 403 response.
/// If the host does not match then the caller is treated as a public non-authenticated caller.
/// </summary>
/// <returns>The caller's SIP account or null for non-hosted domains.</returns>
private async Task <ISIPAccount> GetCaller(UASInviteTransaction uasTx)
{
var invReq = uasTx.TransactionRequest;
if (invReq.Header.From == null || invReq.Header.From.FromURI == null)
{
uasTx.SendFinalResponse(SIPResponse.GetResponse(invReq, SIPResponseStatusCodesEnum.BadRequest, "From header malformed"));
return(null);
}
else
{
string canonicalDomain = _sipDomainManager.GetCanonicalDomain(invReq.Header.From.FromURI.HostAddress);
if (canonicalDomain == null)
{
// The caller is from a non-hosted domain. Will be treated as a public non-authenticated caller.
return(null);
}
else
{
Logger.LogDebug($"B2B incoming caller was for hosted domain {canonicalDomain}, looking up caller for {invReq.Header.From.FromURI.User}.");
var sipAccount = await _sipAccountDataLayer.GetSIPAccount(invReq.Header.From.FromURI.User, canonicalDomain);
if (sipAccount == null)
{
Logger.LogWarning($"B2B no SIP account found for caller {invReq.Header.From.FromURI.User}@{canonicalDomain}, rejecting.");
uasTx.SendFinalResponse(SIPResponse.GetResponse(invReq, SIPResponseStatusCodesEnum.Forbidden, null));
OnAcceptCallFailure?.Invoke(uasTx.TransactionRequest.RemoteSIPEndPoint, CallFailureEnum.NoSIPAccount, invReq);
}
return(sipAccount);
}
}
}
public void Start(string commandData)
{
try
{
m_rtspAppStruct = new RTSPAppStruct(commandData, false);
RTSPClient rtspClient = new RTSPClient();
string sdp = rtspClient.GetStreamDescription(m_rtspAppStruct.URL);
SIPResponse sipResponse = GetOkResponse(m_clientTransaction.TransactionRequest, sdp);
m_clientTransaction.SendFinalResponse(sipResponse);
//rtspClient.Start(m_rtspAppStruct.URL, m_clientTransaction.RemoteEndPoint.Address, m_rtpPort);
}
catch (Exception excp)
{
logger.Error("Exception RTSPCall StartCall. " + excp.Message);
}
}
private void Forward(UASInviteTransaction uasTx)
{
SIPB2BUserAgent b2bua = new SIPB2BUserAgent(_sipTransport, null, uasTx, null);
b2bua.CallAnswered += (uac, resp) => ForwardCallAnswered(uac, b2bua);
var dst = _getDestination(uasTx);
if (dst == null)
{
Logger.LogInformation($"B2BUA lookup did not return a destination. Rejecting UAS call.");
var notFoundResp = SIPResponse.GetResponse(uasTx.TransactionRequest, SIPResponseStatusCodesEnum.NotFound, null);
uasTx.SendFinalResponse(notFoundResp);
}
else
{
Logger.LogInformation($"B2BUA forwarding call to {dst.Uri}.");
b2bua.Call(dst);
}
}
private async Task Forward(UASInviteTransaction uasTx, ISIPAccount callerSIPAccount)
{
var invReq = uasTx.TransactionRequest;
//uasTx.TransactionStateChanged += (tx) => Logger.LogDebug($"B2B uas tx state changed to {tx.TransactionState}.");
//uasTx.TransactionTraceMessage += (tx, msg) => Logger.LogDebug($"B2B uas tx trace message. {msg}");
Logger.LogDebug($"B2B commencing forward for caller {invReq.Header.From.FromURI} to {invReq.URI}.");
SIPB2BUserAgent b2bua = new SIPB2BUserAgent(_sipTransport, null, uasTx, callerSIPAccount);
bool isAuthenticated = false;
if (callerSIPAccount != null)
{
isAuthenticated = b2bua.AuthenticateCall();
}
if (callerSIPAccount == null || isAuthenticated == true)
{
b2bua.CallAnswered += (uac, resp) => ForwardCallAnswered(uac, b2bua);
var dst = await _sipdialPlan.Lookup(uasTx, null);
if (dst == null)
{
Logger.LogInformation($"B2BUA lookup did not return a destination. Rejecting UAS call.");
var notFoundResp = SIPResponse.GetResponse(uasTx.TransactionRequest, SIPResponseStatusCodesEnum.NotFound, null);
uasTx.SendFinalResponse(notFoundResp);
OnAcceptCallFailure?.Invoke(uasTx.TransactionRequest.RemoteSIPEndPoint, CallFailureEnum.NotFound, invReq);
}
else
{
Logger.LogInformation($"B2BUA forwarding call to {dst.Uri}.");
b2bua.Call(dst);
}
}
}
static void Main()
{
Console.WriteLine("SIPSorcery Attended Transfer example.");
Console.WriteLine("Press 'c' to place a call to the default destination.");
Console.WriteLine("Place two simultaneous SIP calls to this program and then press 't'.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
// Create two user agents. Each gets configured to answer an incoming call.
var userAgent1 = new SIPUserAgent(sipTransport, null);
var userAgent2 = new SIPUserAgent(sipTransport, null);
userAgent1.OnCallHungup += (dialog) => Log.LogInformation($"UA1: Call hungup by remote party.");
userAgent1.ServerCallCancelled += (uas) => Log.LogInformation("UA1: Incoming call cancelled by caller.");
userAgent2.OnCallHungup += (dialog) => Log.LogInformation($"UA2: Call hungup by remote party.");
userAgent2.ServerCallCancelled += (uas) => Log.LogInformation("UA2: Incoming call cancelled by caller.");
userAgent2.OnTransferNotify += (sipFrag) =>
{
if (!string.IsNullOrEmpty(sipFrag))
{
Log.LogInformation($"UA2: Transfer status update: {sipFrag.Trim()}.");
if (sipFrag?.Contains("SIP/2.0 200") == true)
{
// The transfer attempt got a successful answer. Can hangup the call.
userAgent2.Hangup();
exitCts.Cancel();
}
}
};
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (!userAgent1.IsCallActive || !userAgent2.IsCallActive)
{
SIPUserAgent activeAgent = (!userAgent1.IsCallActive) ? userAgent1 : userAgent2;
string agentDesc = (!userAgent1.IsCallActive) ? "UA1" : "UA2";
Log.LogInformation($"{agentDesc}: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = activeAgent.AcceptCall(sipRequest);
var rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await activeAgent.Answer(incomingCall, rtpAVSession);
Log.LogInformation($"{agentDesc}: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
else
{
// If both user agents are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
// Place an outgoing call using the first free user agent.
SIPUserAgent freeAgent = (!userAgent1.IsCallActive) ? userAgent1 : (!userAgent2.IsCallActive) ? userAgent2 : null;
if (freeAgent != null)
{
var rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await freeAgent.Call(DEFAULT_DESTINATION_SIP_URI, null, null, rtpAVSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("Both user agents are already on calls.");
}
}
if (keyProps.KeyChar == 't')
{
// Initiate the attended transfer.
if (userAgent1.IsCallActive && userAgent2.IsCallActive)
{
bool result = await userAgent2.AttendedTransfer(userAgent1.Dialogue, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (!result)
{
Log.LogWarning($"Attended transfer failed.");
}
}
else
{
Log.LogWarning("There need to be two active calls before the attended transfer can occur.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
userAgent1?.Hangup();
userAgent2?.Hangup();
// Give any BYE or CANCEL requests time to be transmitted.
Log.LogInformation("Waiting 1s for calls to be cleaned up...");
Task.Delay(1000).Wait();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
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)
{
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);
if (authenticationResult.Authenticated)
{
if (authenticationResult.WasAuthenticatedByIP)
{
}
else
{
}
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;
m_uasTransaction.SendFinalResponse(authReqdResponse);
}
}
}
}
catch (Exception excp)
{
Logger.Logger.Error("Exception SIPServerUserAgent AuthenticateCall. ->" + excp.Message);
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
return(m_isAuthenticated);
}
/// <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="sipRequest">The in dialog request received.</param>
private async Task DialogRequestReceivedAsync(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 = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
var sendResult = await SendResponseAsync(noCallLegResponse);
if (sendResult != SocketError.Success)
{
logger.LogWarning($"SIPUserAgent send response failed in DialogRequestReceivedAsync with {sendResult}.");
}
}
else
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
logger.LogInformation($"Remote call party hungup {sipRequest.StatusLine}.");
Dialogue.DialogueState = SIPDialogueStateEnum.Terminated;
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await SendResponseAsync(okResponse).ConfigureAwait(false);
CallEnded();
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
logger.LogDebug($"Re-INVITE request received {sipRequest.StatusLine}.");
UASInviteTransaction reInviteTransaction = new UASInviteTransaction(m_transport, sipRequest, m_outboundProxy);
try
{
var answerSdp = await MediaSession.RemoteReInvite(sipRequest.Body).ConfigureAwait(false);
Dialogue.RemoteSDP = sipRequest.Body;
Dialogue.SDP = answerSdp;
Dialogue.RemoteCSeq = sipRequest.Header.CSeq;
var okResponse = reInviteTransaction.GetOkResponse(SDP.SDP_MIME_CONTENTTYPE, Dialogue.SDP);
reInviteTransaction.SendFinalResponse(okResponse);
}
catch (Exception ex)
{
logger.LogError(ex, "MediaSession can't process the re-INVITE request.");
if (OnReinviteRequest == null)
{
// The application isn't prepared to accept re-INVITE requests and we can't work out what it was for.
// We'll reject as gently as we can to try and not lose the call.
SIPResponse notAcceptableResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptable, null);
reInviteTransaction.SendFinalResponse(notAcceptableResponse);
}
else
{
// The application is going to handle the re-INVITE request. We'll send a Trying response as a precursor.
SIPResponse tryingResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
await reInviteTransaction.SendProvisionalResponse(tryingResponse);
OnReinviteRequest.Invoke(reInviteTransaction);
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "OPTIONS request for established dialogue " + dialogue.DialogueName + ".", dialogue.Owner));
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Body = Dialogue.RemoteSDP;
okResponse.Header.ContentLength = okResponse.Body.Length;
okResponse.Header.ContentType = m_sdpContentType;
await SendResponseAsync(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));
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await m_transport.SendResponseAsync(okResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REFER)
{
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 = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing mandatory Refer-To header");
await SendResponseAsync(invalidResponse);
}
else
{
//TODO: Add handling logic for in transfer requests from the remote call party.
}
}
else if (sipRequest.Method == SIPMethodsEnum.NOTIFY)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await SendResponseAsync(okResponse);
if (sipRequest.Body?.Length > 0 && sipRequest.Header.ContentType?.Contains(m_sipReferContentType) == true)
{
OnTransferNotify?.Invoke(sipRequest.Body);
}
}
}
}
//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.
Log = 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($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}...");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//EnableTraceLogs(sipTransport);
RTPSession rtpSession = null;
// Create a SIP user agent to receive a call from a remote SIP client.
// Wire up event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
// We're only answering SIP calls, not placing them.
userAgent.OnCallHungup += (dialog) =>
{
Log.LogInformation($"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);
rtpSession = new RTPSession(false, false, false);
rtpSession.AcceptRtpFromAny = true;
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
await userAgent.Answer(incomingCall, rtpSession);
rtpSession.OnRtpPacketReceived += (ep, 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...");
rtpSession?.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();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
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(m_sipUsername, 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 =
SIPResponse.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.LogError("Exception SIPServerUserAgent AuthenticateCall. " + excp.Message);
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
return(m_isAuthenticated);
}
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.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
EnableTraceLogs(sipTransport);
// Create two user agents. Each gets configured to answer an incoming call.
var userAgent1 = new SIPUserAgent(sipTransport, null);
var userAgent2 = new SIPUserAgent(sipTransport, null);
// Only one of the user agents can use the microphone and speaker. The one designated
// as the active agent gets the devices.
SIPUserAgent activeUserAgent = null;
RTPMediaSession activeRtpSession = null;
// Get the default speaker.
var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice();
m_audioOutProvider = audioOutProvider;
WaveInEvent waveInEvent = GetAudioInputDevice();
userAgent1.OnCallHungup += () => Log.LogInformation($"UA1: Call hungup by remote party.");
userAgent1.ServerCallCancelled += (uas) => Log.LogInformation("UA1: Incoming call cancelled by caller.");
userAgent2.OnCallHungup += () => Log.LogInformation($"UA2: Call hungup by remote party.");
userAgent2.ServerCallCancelled += (uas) => Log.LogInformation("UA2: Incoming call cancelled by caller.");
userAgent2.OnTransferNotify += (sipFrag) =>
{
if (!string.IsNullOrEmpty(sipFrag))
{
Log.LogInformation($"UA2: Transfer status update: {sipFrag.Trim()}.");
if (sipFrag?.Contains("SIP/2.0 200") == true)
{
// The transfer attempt got a succesful answer. Can hangup the call.
userAgent2.Hangup();
exitCts.Cancel();
}
}
};
sipTransport.SIPTransportRequestReceived += (locelEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (!userAgent1.IsCallActive)
{
Log.LogInformation($"UA1: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent1.AcceptCall(sipRequest);
var rtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, new SDPMediaFormat(SDPMediaFormatsEnum.PCMU), AddressFamily.InterNetwork);
rtpMediaSession.RemotePutOnHold += () => Log.LogInformation("UA1: Remote call party has placed us on hold.");
rtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("UA1: Remote call party took us off hold.");
userAgent1.Answer(incomingCall, rtpMediaSession)
.ContinueWith(task =>
{
activeUserAgent = userAgent1;
activeRtpSession = rtpMediaSession;
activeRtpSession.OnRtpPacketReceived += PlaySample;
waveInEvent.StartRecording();
Log.LogInformation($"UA1: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}, exitCts.Token);
}
else if (!userAgent2.IsCallActive)
{
Log.LogInformation($"UA2: Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent2.AcceptCall(sipRequest);
var rtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, new SDPMediaFormat(SDPMediaFormatsEnum.PCMU), AddressFamily.InterNetwork);
rtpMediaSession.RemotePutOnHold += () => Log.LogInformation("UA2: Remote call party has placed us on hold.");
rtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("UA2: Remote call party took us off hold.");
userAgent2.Answer(incomingCall, rtpMediaSession)
.ContinueWith(task =>
{
activeRtpSession.OnRtpPacketReceived -= PlaySample;
activeUserAgent = userAgent2;
activeRtpSession = rtpMediaSession;
activeRtpSession.PutOnHold();
activeRtpSession.OnRtpPacketReceived += PlaySample;
Log.LogInformation($"UA2: Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}, exitCts.Token);
}
else
{
// If both user agents are already on a call return a busy response.
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
return(sipTransport.SendResponseAsync(notAllowedResponse));
}
return(Task.FromResult(0));
};
// Wire up the RTP send session to the audio input 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 (activeRtpSession != null)
{
activeRtpSession.SendAudioFrame(rtpSendTimestamp, (int)SDPMediaFormatsEnum.PCMU, sample);
rtpSendTimestamp += (uint)sample.Length;
}
};
// 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 == 't')
{
if (userAgent1.IsCallActive && userAgent2.IsCallActive)
{
bool result = await userAgent2.AttendedTransfer(userAgent1.Dialogue, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (!result)
{
Log.LogWarning($"Attended transfer failed.");
}
}
else
{
Log.LogWarning("There need to be two active calls before the attended transfer can occur.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
userAgent1?.Hangup();
userAgent2?.Hangup();
waveInEvent?.StopRecording();
audioOutEvent?.Stop();
// 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
}
static void Main()
{
Console.WriteLine("SIPSorcery Call Hold and Blind Transfer example.");
Console.WriteLine("Press 'c' to initiate a call to the default destination.");
Console.WriteLine("Press 'h' to place an established call on and off hold.");
Console.WriteLine("Press 'H' to hangup an established call.");
Console.WriteLine("Press 't' to request a blind transfer on an established call.");
Console.WriteLine("Press 'q' or ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
Console.WriteLine($"Listening for incoming calls on: {sipTransport.GetSIPChannels().First().ListeningEndPoint}.");
EnableTraceLogs(sipTransport);
_currentDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
RtpAVSession rtpAVSession = null;
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold.");
userAgent.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold.");
sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) =>
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (userAgent?.IsCallActive == true)
{
Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
// If we are already on a call return a busy response.
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null);
uasTransaction.SendFinalResponse(busyResponse);
}
else
{
Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}.");
var incomingCall = userAgent.AcceptCall(sipRequest);
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
await userAgent.Answer(incomingCall, rtpAVSession);
Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}.");
}
}
else
{
Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowedResponse);
}
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'c')
{
if (!userAgent.IsCallActive)
{
rtpAVSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
bool callResult = await userAgent.Call(DEFAULT_DESTINATION_SIP_URI, SIP_USERNAME, SIP_PASSWORD, rtpAVSession);
Log.LogInformation($"Call attempt {((callResult) ? "successfull" : "failed")}.");
}
else
{
Log.LogWarning("There is already an active call.");
}
}
else if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsCallActive)
{
if (userAgent.IsOnLocalHold)
{
Log.LogInformation("Taking the remote call party off hold.");
userAgent.TakeOffHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions {
AudioSource = AudioSourcesEnum.CaptureDevice
}, null);
}
else
{
Log.LogInformation("Placing the remote call party on hold.");
userAgent.PutOnHold();
await(userAgent.MediaSession as RtpAVSession).SetSources(new AudioOptions
{
AudioSource = AudioSourcesEnum.Music,
SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>
{
{ SDPMediaFormatsEnum.PCMU, _currentDir + "/" + AUDIO_FILE_PCMU }
}
}, null);
}
}
else
{
Log.LogWarning("There is no active call to put on hold.");
}
}
else if (keyProps.KeyChar == 'H')
{
if (userAgent.IsCallActive)
{
Log.LogInformation("Hanging up call.");
userAgent.Hangup();
}
}
else if (keyProps.KeyChar == 't')
{
// Initiate a blind transfer to the remote call party.
if (userAgent.IsCallActive)
{
var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI);
bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (result)
{
// If the transfer was accepted the original call will already have been hungup.
// Wait a second for the transfer NOTIFY request to arrive.
await Task.Delay(1000);
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else
{
Log.LogWarning("There is no active call to transfer.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
rtpAVSession?.Close("app exit");
if (userAgent != null)
{
if (userAgent.IsCallActive)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
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;
}
}
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($"Cancelling call to {userAgent?.CallDescriptor?.To}.");
userAgent.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
#endregion
}
/// <summary>
/// Handler for when an in dialog request is received on an established call.
/// Typical types of request will be re-INVITES for things like putting a call on or
/// off hold and REFER requests for transfers. Some in dialog request types, such
/// as re-INVITES have specific events so they can be bubbled up to the
/// application to deal with.
/// </summary>
/// <param name="sipRequest">The in dialog request received.</param>
private async Task DialogRequestReceivedAsync(SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
logger.LogInformation($"Remote call party hungup {sipRequest.StatusLine}.");
Dialogue.DialogueState = SIPDialogueStateEnum.Terminated;
SIPNonInviteTransaction byeTx = new SIPNonInviteTransaction(m_transport, sipRequest, null);
byeTx.SendResponse(SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null));
CallEnded();
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
logger.LogDebug($"Re-INVITE request received {sipRequest.StatusLine}.");
UASInviteTransaction reInviteTransaction = new UASInviteTransaction(m_transport, sipRequest, m_outboundProxy);
try
{
MediaSession.setRemoteDescription(new RTCSessionDescription {
sdp = SDP.ParseSDPDescription(sipRequest.Body), type = RTCSdpType.offer
});
CheckRemotePartyHoldCondition(MediaSession.remoteDescription.sdp);
var answerSdp = await MediaSession.createAnswer(null).ConfigureAwait(false);
Dialogue.RemoteSDP = sipRequest.Body;
Dialogue.SDP = answerSdp.ToString();
Dialogue.RemoteCSeq = sipRequest.Header.CSeq;
var okResponse = reInviteTransaction.GetOkResponse(SDP.SDP_MIME_CONTENTTYPE, Dialogue.SDP);
reInviteTransaction.SendFinalResponse(okResponse);
}
catch (Exception ex)
{
logger.LogError(ex, "MediaSession can't process the re-INVITE request.");
if (OnReinviteRequest == null)
{
// The application isn't prepared to accept re-INVITE requests and we can't work out what it was for.
// We'll reject as gently as we can to try and not lose the call.
SIPResponse notAcceptableResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptable, null);
reInviteTransaction.SendFinalResponse(notAcceptableResponse);
}
else
{
// The application is going to handle the re-INVITE request. We'll send a Trying response as a precursor.
SIPResponse tryingResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
await reInviteTransaction.SendProvisionalResponse(tryingResponse).ConfigureAwait(false);
OnReinviteRequest.Invoke(reInviteTransaction);
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "OPTIONS request for established dialogue " + dialogue.DialogueName + ".", dialogue.Owner));
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Body = Dialogue.RemoteSDP;
okResponse.Header.ContentLength = okResponse.Body.Length;
okResponse.Header.ContentType = m_sdpContentType;
await SendResponseAsync(okResponse).ConfigureAwait(false);
}
else if (sipRequest.Method == SIPMethodsEnum.MESSAGE)
{
//Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "MESSAGE for call " + sipRequest.URI.ToString() + ": " + sipRequest.Body + ".", dialogue.Owner));
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await m_transport.SendResponseAsync(okResponse).ConfigureAwait(false);
}
else if (sipRequest.Method == SIPMethodsEnum.REFER)
{
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 = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing mandatory Refer-To header");
await SendResponseAsync(invalidResponse).ConfigureAwait(false);
}
else
{
//TODO: Add handling logic for in transfer requests from the remote call party.
}
}
else if (sipRequest.Method == SIPMethodsEnum.NOTIFY)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await SendResponseAsync(okResponse).ConfigureAwait(false);
if (sipRequest.Body?.Length > 0 && sipRequest.Header.ContentType?.Contains(m_sipReferContentType) == true)
{
OnTransferNotify?.Invoke(sipRequest.Body);
}
}
}
/// <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.
}
}
}
}