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);
}
}
}
public void AckRecognitionUnitTest()
{
SIPTransport clientTransport = null;
SIPTransport serverTransport = null;
try
{
SIPTransactionEngine clientEngine = new SIPTransactionEngine(); // Client side of the INVITE.
SIPEndPoint clientEndPoint =
new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 12013));
clientTransport =
new SIPTransport(MockSIPDNSManager.Resolve, clientEngine, new SIPUDPChannel(clientEndPoint.GetIPEndPoint()), false);
SetTransportTraceEvents(clientTransport);
SIPTransactionEngine serverEngine = new SIPTransactionEngine(); // Server side of the INVITE.
UASInviteTransaction serverTransaction = null;
SIPEndPoint serverEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 12014));
serverTransport =
new SIPTransport(MockSIPDNSManager.Resolve, serverEngine, new SIPUDPChannel(serverEndPoint.GetIPEndPoint()), false);
SetTransportTraceEvents(serverTransport);
serverTransport.SIPTransportRequestReceived += (localEndPoint, remoteEndPoint, sipRequest) =>
{
Console.WriteLine("Server Transport Request In: " + sipRequest.Method + ".");
serverTransaction =
serverTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localEndPoint, null);
SetTransactionTraceEvents(serverTransaction);
serverTransaction.GotRequest(localEndPoint, remoteEndPoint, sipRequest);
};
SIPURI dummyURI = SIPURI.ParseSIPURI("sip:dummy@" + serverEndPoint);
SIPRequest inviteRequest = GetDummyINVITERequest(dummyURI);
inviteRequest.LocalSIPEndPoint = clientTransport.GetDefaultTransportContact(SIPProtocolsEnum.udp);
// Send the invite to the server side.
UACInviteTransaction clientTransaction =
new UACInviteTransaction(clientTransport, inviteRequest, serverEndPoint, clientEndPoint, null);
SetTransactionTraceEvents(clientTransaction);
clientEngine.AddTransaction(clientTransaction);
clientTransaction.SendInviteRequest(serverEndPoint, inviteRequest);
Thread.Sleep(500);
Assert.IsTrue(clientTransaction.TransactionState == SIPTransactionStatesEnum.Completed, "Client transaction in incorrect state.");
Assert.IsTrue(serverTransaction.TransactionState == SIPTransactionStatesEnum.Confirmed, "Server transaction in incorrect state.");
}
finally
{
if (clientTransport != null)
{
clientTransport.Shutdown();
}
if (serverTransport != null)
{
serverTransport.Shutdown();
}
}
}
public void Call(SIPCallDescriptor sipCallDescriptor)
{
try
{
m_uacCallDescriptor = sipCallDescriptor;
SIPRequest uacInviteRequest = GetInviteRequest(m_uacCallDescriptor.Uri, sipCallDescriptor);
if (sipCallDescriptor.MangleResponseSDP && sipCallDescriptor.MangleIPAddress != null)
{
uacInviteRequest.Header.ProxyReceivedFrom = sipCallDescriptor.MangleIPAddress.ToString();
}
uacInviteRequest.Body = sipCallDescriptor.Content;
uacInviteRequest.Header.ContentType = sipCallDescriptor.ContentType;
uacInviteRequest.LocalSIPEndPoint = m_blackhole;
uacInviteRequest.RemoteSIPEndPoint = m_blackhole;
// Now that we have a destination socket create a new UAC transaction for forwarded leg of the call.
m_uacTransaction =
m_sipTransport.CreateUACTransaction(uacInviteRequest, m_blackhole, m_blackhole, null);
if (m_uacTransaction.CDR != null)
{
m_uacTransaction.CDR.Owner = m_uacOwner;
m_uacTransaction.CDR.AdminMemberId = m_uacAdminMemberId;
m_uacTransaction.CDR.DialPlanContextID = (m_uacCallDescriptor != null)
? m_uacCallDescriptor.DialPlanContextID
: Guid.Empty;
}
//uacTransaction.UACInviteTransactionInformationResponseReceived += ServerInformationResponseReceived;
//uacTransaction.UACInviteTransactionFinalResponseReceived += ServerFinalResponseReceived;
//uacTransaction.UACInviteTransactionTimedOut += ServerTimedOut;
//uacTransaction.TransactionTraceMessage += TransactionTraceMessage;
m_uacTransaction.SendInviteRequest(m_blackhole, m_uacTransaction.TransactionRequest);
SIPRequest uasInviteRequest = uacInviteRequest.Copy();
uasInviteRequest.LocalSIPEndPoint = m_blackhole;
uasInviteRequest.RemoteSIPEndPoint = m_blackhole;
uasInviteRequest.Header.Vias.TopViaHeader.Branch = CallProperties.CreateBranchId();
m_uasTransaction =
m_sipTransport.CreateUASTransaction(uasInviteRequest, m_blackhole, m_blackhole, null);
SetOwner(sipCallDescriptor.ToSIPAccount.Owner, sipCallDescriptor.ToSIPAccount.AdminMemberId);
//m_uasTransaction.TransactionTraceMessage += TransactionTraceMessage;
//m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
//m_uasTransaction.UASInviteTransactionCancelled += (t) => { };
QueueNewCall_External(this);
CallTrying.Invoke(this, null);
}
catch (Exception excp)
{
logger.Error("Exception SIPB2BUserAgent Call. " + excp.Message);
}
}
/// <summary>
/// 发送可靠的sip请求消息
/// </summary>
/// <param name="remoteEP">远程终结点</param>
/// <param name="request">sip请求消息</param>
public void SendReliableRequest(SIPEndPoint remoteEP, SIPRequest request)
{
if (_serviceState == ServiceStatus.Wait)
{
OnSIPServiceChange(remoteEP.ToHost(), ServiceStatus.Wait);
return;
}
SIPTransaction trans = Transport.CreateUASTransaction(request, remoteEP, LocalEP, null);
trans.TransactionStateChanged += trans_TransactionStateChanged;
trans.SendReliableRequest();
}
private DialPlanLineContext GetDummyDialPlanContext(string testDialPlan, string dst)
{
SIPDialPlan dialPlan = new SIPDialPlan(null, null, null, testDialPlan, SIPDialPlanScriptTypesEnum.Asterisk);
SIPTransactionEngine transactionEngine = new SIPTransactionEngine();
SIPTransport sipTransport = new SIPTransport(MockSIPDNSManager.Resolve, transactionEngine);
SIPURI dummyURI = SIPURI.ParseSIPURI(dst);
SIPRequest inviteRequest = GetDummyINVITERequest(dummyURI);
SIPEndPoint dummyEndPoint = SIPEndPoint.ParseSIPEndPoint("udp:0.0.0.0:5060");
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(inviteRequest, dummyEndPoint, dummyEndPoint, null);
SIPServerUserAgent uas = new SIPServerUserAgent(sipTransport, null, "test", "sipsorcery.com", SIPCallDirection.In, null, null, null, uasTransaction);
DialPlanLineContext dialPlanContext = new DialPlanLineContext(null, null, null, null, uas, dialPlan, null, null, null, null);
return(dialPlanContext);
}
/// <summary>
/// Handler for processing incomign 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.");
m_uas.SIPDialogue.Hangup(m_sipTransport, null);
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;
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);
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2("localhost.pfx");
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.G722)))
{
Log.LogDebug($"Using G722 RTP media type and audio file {AUDIO_FILE_G722}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.G722, null, null);
audioFile = AUDIO_FILE_G722;
}
else if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(dstRtpEndPoint.AddressFamily == AddressFamily.InterNetworkV6 ? IPAddress.IPv6Any : IPAddress.Any, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
IPEndPoint rtpEndPoint = new IPEndPoint(rtpAddress, (rtpSocket.LocalEndPoint as IPEndPoint).Port);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, audioFile, rtpCts))
.ContinueWith(_ =>
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
private static readonly int RTP_REPORTING_PERIOD_SECONDS = 5; // Period at which to write RTP stats.
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
IPAddress defaultAddr = LocalIPConfig.GetDefaultIPv4Address();
var sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new SIPTransactionEngine());
int port = FreePort.FindNextAvailableUDPPort(SIPConstants.DEFAULT_SIP_PORT);
var sipChannel = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));
sipTransport.AddSIPChannel(sipChannel);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource uasCts = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept a little it simpler.
uas?.Hangup();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uasCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(defaultAddr, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, uasCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup();
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup();
uasCts?.Cancel();
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
if (uas?.IsHungup == false)
{
uas?.Hangup();
}
uasCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
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);
}
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to 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)));
// Un/comment this line to see/hide each SIP message sent and received.
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPUserAgent userAgent = new SIPUserAgent(sipTransport, null);
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
}
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (userAgent?.IsAnswered == true)
{
userAgent?.Hangup();
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
if (userAgent.AcceptCall(uasTransaction))
{
rtpCts = new CancellationTokenSource();
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(rtpAddress, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
rtpSendSession.DestinationEndPoint = dstRtpEndPoint;
rtpRecvSession.OnReceiveFromEndPointChanged += (oldEP, newEP) =>
{
Log.LogDebug($"RTP destination end point changed from {oldEP} to {newEP}.");
rtpSendSession.DestinationEndPoint = newEP;
};
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
userAgent.Answer(GetSDP(rtpSocket.LocalEndPoint as IPEndPoint));
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
rtpCts?.Cancel();
};
// 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')
{
// Initiate a transfer.
bool transferResult = await userAgent.Transfer(SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI), new TimeSpan(0, 0, TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (transferResult)
{
// If the transfer was accepted the original call will already have been hungup.
userAgent = null;
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
Log.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (userAgent != null)
{
if (userAgent.IsAnswered)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the final request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.ContactHost = Dns.GetHostName();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//if (File.Exists("localhost.pfx"))
//{
// var certificate = new X509Certificate2(@"localhost.pfx", "");
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.Any, SIPS_LISTEN_PORT)));
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.IPv6Any, SIPS_LISTEN_PORT)));
//}
//EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
// In practice there could be a number of reasons to reject the call. Unsupported extensions, mo matching codecs etc. etc.
if (sipRequest.Header.HasUnknownRequireExtension)
{
// The caller requires an extension that we don't support.
SIPSorcery.Sys.Log.Logger.LogWarning($"Rejecting incoming call due to one or more required exensions not being supported, required extensions: {sipRequest.Header.Require}.");
uas.Reject(SIPResponseStatusCodesEnum.NotImplemented, "Unsupported Require Extension", null);
}
else
{
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Simulating answer delay to test provisional response retransmits.
await Task.Delay(2000);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localSIPEndPoint.Address, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, rtpCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
try
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{localSIPEndPoint.ToString()}<-{remoteEndPoint.ToString()}: {sipRequest.StatusLine}");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
catch (Exception optionsExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Failed to send SIP OPTIONS response. {optionsExcp.Message}");
}
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpCts?.Cancel();
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to hangup...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
/// <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.
}
}
}
}
private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
var endpoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (transaction != null)
{
return;
}
logger.DebugFormat("{0} Incoming call from {1}", prefix, sipRequest.Header.From.FromURI.User);
transaction = transport.CreateUASTransaction(sipRequest, remoteEndPoint, endpoint, null);
agent = new SIPServerUserAgent(
transport,
null,
sipRequest.Header.From.FromURI.User,
null,
SIPCallDirection.In,
null,
null,
null,
transaction);
agent.CallCancelled += Agent_CallCancelled;
agent.TransactionComplete += Agent_TransactionComplete;
agent.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
agent.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answer = SDP.ParseSDPDescription(agent.CallRequest.Body);
var address = IPAddress.Parse(answer.Connection.ConnectionAddress);
var port = answer.Media.FirstOrDefault(m => m.Media == SDPMediaTypesEnum.audio).Port;
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(address, port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
// TODO Fix hardcoded ports
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += Channel_OnFrameReady;
rtpChannel.Start();
// Send some setup parameters to punch a hole in the firewall/router
rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });
rtpChannel.OnControlDataReceived += (b) => { logger.Debug($"{prefix} Control Data Received; {b.Length} bytes"); };
rtpChannel.OnControlSocketDisconnected += () => { logger.Debug($"{prefix} Control Socket Disconnected"); };
var announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
},
Port = rtpChannel.RTPPort
};
sdp.Media.Add(announcement);
SetState(State.Listening, sipRequest.Header.From.FromURI.User);
agent.Progress(SIPResponseStatusCodesEnum.Accepted, null, null, null, null);
agent.Answer(SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
SetState(State.Busy, "");
return;
}
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (State != State.Busy)
{
return;
}
logger.DebugFormat("{0} Hangup from {1}", prefix, sipRequest.Header.From.FromURI.User);
var noninvite = transport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, endpoint, null);
var response = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
noninvite.SendFinalResponse(response);
SetState(State.Finished, Endpoint);
rtpChannel.OnFrameReady -= Channel_OnFrameReady;
rtpChannel.Close();
agent.TransactionComplete -= Agent_TransactionComplete;
agent.CallCancelled -= Agent_CallCancelled;
agent = null;
transaction = null;
SetState(State.Ready, Endpoint);
return;
}
if (sipRequest.Method == SIPMethodsEnum.ACK)
{
}
if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
}
}
