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 trnasport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
EnableTraceLogs(sipTransport);
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
// Only set the remote RTP end point if there hasn't already been a packet received on it.
if (_remoteRtpEndPoint == null)
{
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
exitCts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, exitCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, exitCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri.ToString(),
$"sip:{SIP_USERNAME}@localhost",
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h')
{
}
else if (keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
uac.Hangup();
exitCts.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"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 (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
/// <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 static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
throw new NotImplementedException();
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
throw new NotImplementedException();
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
throw new NotImplementedException();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
SIPNonInviteTransaction optionsTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
optionsTransaction.SendFinalResponse(optionsResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponseStatusCodesEnum registerResponse = SIPResponseStatusCodesEnum.Ok;
if (sipRequest.Header.Contact != null && sipRequest.Header.Contact.Count > 0)
{
int expiry = sipRequest.Header.Contact[0].Expires > 0 ? sipRequest.Header.Contact[0].Expires : sipRequest.Header.Expires;
var sipAccount = new SIPAccount(null, sipRequest.Header.From.FromURI.Host, sipRequest.Header.From.FromURI.User, null, null);
SIPRegistrarBinding binding = new SIPRegistrarBinding(sipAccount, sipRequest.Header.Contact[0].ContactURI, null, 0, null, remoteEndPoint, localSIPEndPoint, null, expiry);
if (_sipRegistrations.ContainsKey(sipAccount.SIPUsername))
{
_sipRegistrations.Remove(sipAccount.SIPUsername);
}
_sipRegistrations.Add(sipAccount.SIPUsername, binding);
logger.Debug("Registered contact for " + sipAccount.SIPUsername + " as " + binding.ToContactString() + ".");
}
else
{
registerResponse = SIPResponseStatusCodesEnum.BadRequest;
}
SIPNonInviteTransaction registerTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, registerResponse, null);
registerTransaction.SendFinalResponse(okResponse);
}
else
{
logger.Debug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
}
}
catch (NotImplementedException)
{
logger.Debug(sipRequest.Method + " request processing not implemented for " + sipRequest.URI.ToParameterlessString() + " from " + remoteEndPoint + ".");
SIPNonInviteTransaction notImplTransaction = _sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse notImplResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotImplemented, null);
notImplTransaction.SendFinalResponse(notImplResponse);
}
}
private static ConcurrentQueue <RTPEvent> _dtmfEvents = new ConcurrentQueue <RTPEvent>(); // Add a DTMF event to this queue to have the it sent
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port)));
// Uncomment this line to see each SIP message sent and received.
EnableTraceLogs(sipTransport);
// Send an OPTIONS request to determine the local IP address to use for the RTP socket.
var optionsTask = SendOptionsTaskAsync(sipTransport, callUri);
var result = Task.WhenAny(optionsTask, Task.Delay(SIP_REQUEST_TIMEOUT_MILLISECONDS));
result.Wait();
if (optionsTask.IsCompletedSuccessfully == false || optionsTask.Result == null)
{
Log.LogError($"OPTIONS request to {callUri} failed.");
}
else
{
IPAddress localIPAddress = optionsTask.Result;
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint, RTPPayloadTypesEnum.PCMU).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
Task.Delay(3000).Wait();
// Add some DTMF events to the queue. These will be transmitted by the SendRtp thread.
_dtmfEvents.Enqueue(new RTPEvent(0x05, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID));
Task.Delay(2000, rtpCts.Token).Wait();
_dtmfEvents.Enqueue(new RTPEvent(0x09, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID));
Task.Delay(2000, rtpCts.Token).Wait();
_dtmfEvents.Enqueue(new RTPEvent(0x02, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID));
Task.Delay(2000, rtpCts.Token).Wait();
Log.LogInformation("Exiting...");
rtpCts.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static 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();
}
};
}
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 " + m_transaction.TransactionRequest.Method + " for " + m_sipUsername + "@" + m_sipDomain + ", SIP account not found.", null));
Reject(SIPResponseStatusCodesEnum.Forbidden, null, null);
}
else
{
SIPRequest sipRequest = m_transaction.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, m_transaction.TransactionRequest.Method + " request from " + remoteEndPoint.ToString() + " successfully authenticated by IP address.", m_sipAccount.Owner));
}
else
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, m_transaction.TransactionRequest.Method + " request from " + remoteEndPoint.ToString() + " successfully authenticated by digest.", m_sipAccount.Owner));
}
SetOwner(m_sipAccount.Owner, m_sipAccount.AdminMemberId);
m_isAuthenticated = true;
}
else
{
// Send authorisation failure or required response
SIPResponse authReqdResponse = SIPTransport.GetResponse(sipRequest, authenticationResult.ErrorResponse, null);
authReqdResponse.Header.AuthenticationHeader = authenticationResult.AuthenticationRequiredHeader;
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, m_transaction.TransactionRequest.Method + " request not authenticated for " + m_sipUsername + "@" + m_sipDomain + ", responding with " + authenticationResult.ErrorResponse + ".", null));
m_transaction.SendFinalResponse(authReqdResponse);
}
}
}
}
catch (Exception excp)
{
logger.LogError("Exception SIPNonInviteUserAgent AuthenticateCall. " + excp.Message);
Reject(SIPResponseStatusCodesEnum.InternalServerError, null, null);
}
return(m_isAuthenticated);
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
else
{
callUri = SIPURI.ParseSIPURIRelaxed(args[0]);
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
IPAddress localAddress = sipTransport.GetLocalAddress(IPAddress.Parse("8.8.8.8"));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(localAddress, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, port)));
//EnableTraceLogs(sipTransport);
// Select the IP address to use for RTP based on the destination SIP URI.
var endPointForCall = callUri.ToSIPEndPoint() == null ? sipTransport.GetDefaultSIPEndPoint(callUri.Protocol) : sipTransport.GetDefaultSIPEndPoint(callUri.ToSIPEndPoint());
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
// TODO (find something better): If the SIP endpoint is using 0.0.0.0 for SIP use loopback for RTP.
IPAddress rtpAddress = localAddress;
NetServices.CreateRtpSocket(rtpAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
rtpCts.Token.WaitHandle.WaitOne();
Log.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
/// <summary>
/// Handler for processing incoming SIP requests.
/// </summary>
/// <param name="localSIPEndPoint">The end point the request was received on.</param>
/// <param name="remoteEndPoint">The end point the request came from.</param>
/// <param name="sipRequest">The SIP request received.</param>
private void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (m_uac != null && m_uac.SIPDialogue != null && sipRequest.Header.CallId == m_uac.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup by remote end.");
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallFinished();
}
else if (m_uas != null && m_uas.SIPDialogue != null && sipRequest.Header.CallId == m_uas.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup.");
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallFinished();
}
else
{
logger.Debug("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
StatusMessage("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
m_uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
m_uas.CallCancelled += UASCallCancelled;
m_uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
m_uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
IncomingCall();
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
StatusMessage("Call was cancelled by remote end.");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
logger.Debug("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
CallFinished();
}
else
{
logger.Debug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
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_transaction.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_transaction.SendFinalResponse(authReqdResponse);
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception SIPNonInviteUserAgent 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="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.
}
}
}
}