public void AddInviteRequest(SIPRequest inviteRequest)
{
if (inviteRequest.Method != SIPMethodsEnum.INVITE)
{
SIPResponse notSupportedResponse = SIPResponse.GetResponse(inviteRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, "Invite requests only");
_sipTransport.SendResponseAsync(notSupportedResponse).Wait();
}
else
{
if (_inviteQueue.Count < MAX_INVITE_QUEUE_SIZE)
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(_sipTransport, inviteRequest, null);
var trying = SIPResponse.GetResponse(inviteRequest, SIPResponseStatusCodesEnum.Trying, null);
uasTransaction.SendProvisionalResponse(trying).Wait();
_inviteQueue.Enqueue(uasTransaction);
}
else
{
Logger.LogWarning($"Invite queue exceeded max queue size {MAX_INVITE_QUEUE_SIZE} overloaded response sent.");
SIPResponse overloadedResponse = SIPResponse.GetResponse(inviteRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, "B2BUA overloaded, please try again shortly");
_sipTransport.SendResponseAsync(overloadedResponse).Wait();
}
_inviteARE.Set();
}
}
/// <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));
}
public SIPServerUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
string sipUsername,
string sipDomain,
SIPCallDirection callDirection,
SIPAssetGetDelegate<SIPAccount> getSIPAccount,
SIPAuthenticateRequestDelegate sipAuthenticateRequest,
SIPMonitorLogDelegate logDelegate,
UASInviteTransaction uasTransaction)
{
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_sipUsername = sipUsername;
m_sipDomain = sipDomain;
m_sipCallDirection = callDirection;
GetSIPAccount_External = getSIPAccount;
SIPAuthenticateRequest_External = sipAuthenticateRequest;
Log_External = logDelegate ?? Log_External;
m_uasTransaction = uasTransaction;
m_uasTransaction.TransactionTraceMessage += TransactionTraceMessage;
m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
m_uasTransaction.UASInviteTransactionCancelled += UASTransactionCancelled;
m_uasTransaction.TransactionRemoved += new SIPTransactionRemovedDelegate(UASTransaction_TransactionRemoved);
//m_uasTransaction.TransactionStateChanged += (t) => { logger.Debug("Transaction state change to " + t.TransactionState + ", uri=" + t.TransactionRequestURI.ToString() + "."); };
}
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 SIPServerUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
string sipUsername,
string sipDomain,
SIPCallDirection callDirection,
GetSIPAccountDelegate getSIPAccount,
SIPAuthenticateRequestDelegate sipAuthenticateRequest,
SIPMonitorLogDelegate logDelegate,
UASInviteTransaction uasTransaction)
{
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_sipUsername = sipUsername;
m_sipDomain = sipDomain;
CallDirection = callDirection;
GetSIPAccount_External = getSIPAccount;
SIPAuthenticateRequest_External = sipAuthenticateRequest;
Log_External = logDelegate ?? Log_External;
m_uasTransaction = uasTransaction;
m_uasTransaction.TransactionTraceMessage += TransactionTraceMessage;
m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
m_uasTransaction.UASInviteTransactionCancelled += UASTransactionCancelled;
m_uasTransaction.TransactionRemoved += new SIPTransactionRemovedDelegate(UASTransaction_TransactionRemoved);
}
public SIPServerUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
string sipUsername,
string sipDomain,
SIPCallDirection callDirection,
SIPAssetGetDelegate <SIPAccount> getSIPAccount,
SIPAuthenticateRequestDelegate sipAuthenticateRequest,
SIPMonitorLogDelegate logDelegate,
UASInviteTransaction uasTransaction)
{
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_sipUsername = sipUsername;
m_sipDomain = sipDomain;
m_sipCallDirection = callDirection;
GetSIPAccount_External = getSIPAccount;
SIPAuthenticateRequest_External = sipAuthenticateRequest;
Log_External = logDelegate ?? Log_External;
m_uasTransaction = uasTransaction;
m_uasTransaction.TransactionTraceMessage += TransactionTraceMessage;
m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
m_uasTransaction.UASInviteTransactionCancelled += UASTransactionCancelled;
m_uasTransaction.TransactionRemoved += new SIPTransactionRemovedDelegate(UASTransaction_TransactionRemoved);
//m_uasTransaction.TransactionStateChanged += (t) => { logger.Debug("Transaction state change to " + t.TransactionState + ", uri=" + t.TransactionRequestURI.ToString() + "."); };
}
public 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);
}
}
public SIPB2BUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
UASInviteTransaction uasTransaction,
ISIPAccount sipAccount) :
base(sipTransport, outboundProxy, uasTransaction, sipAccount)
{
IsB2B = true;
base.CallCancelled += SIPServerUserAgent_CallCancelled;
}
public async Task BlindTransferCancelUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 343" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF +
"Min-SE: 90" + m_CRLF +
"" + m_CRLF +
"v=0" + m_CRLF +
"o=- 3785527268 3785527269 IN IP4 192.168.11.50" + m_CRLF +
"s=pjmedia" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 4032 RTP/AVP 0 101" + m_CRLF +
"c=IN IP4 192.168.11.50" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
SIPServerUserAgent mockUas = new SIPServerUserAgent(transport, null, null, null, SIPCallDirection.In, null, null, null, uasTx);
await userAgent.Answer(mockUas, CreateMediaSession());
CancellationTokenSource cts = new CancellationTokenSource();
var blindTransferTask = userAgent.BlindTransfer(SIPURI.ParseSIPURIRelaxed("127.0.0.1"), TimeSpan.FromSeconds(2), cts.Token);
cts.Cancel();
Assert.False(await blindTransferTask);
//await Assert.ThrowsAnyAsync<TaskCanceledException>(async () => { bool result = ; });
}
public async Task CheckRemoteSocketProxyReceivedUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPEndPoint dummyEP = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 5060));
string inviteReqStr =
@"INVITE sip:[email protected]:12014 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:1234;branch=z9hG4bK5f37455955ca433a902f8fea0ce2dc27
To: <sip:[email protected]:12014>
From: <sip:[email protected]>;tag=2062917371
Call-ID: 8ae45c15425040179a4285d774ccbaf6
CSeq: 1 INVITE
Contact: <sip:127.0.0.1:1234>
Max-Forwards: 70
User-Agent: unittest
Content-Length: 5
Content-Type: application/sdp
Proxy-ReceivedFrom: udp:192.168.0.50:5080
dummy";
var sipReqBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummyEP, dummyEP);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipReqBuffer);
string okRespStr =
@"SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:1234;branch=z9hG4bK5f37455955ca433a902f8fea0ce2dc27;rport=12013
To: <sip:[email protected]:12014>
From: <sip:[email protected]>;tag=2062917371
Call-ID: 8ae45c15425040179a4285d774ccbaf6
CSeq: 1 INVITE
Content-Length: 5
Content-Type: application/sdp
dummy";
var sipRespBuffer = SIPMessageBuffer.ParseSIPMessage(okRespStr, dummyEP, dummyEP);
SIPResponse okResponse = SIPResponse.ParseSIPResponse(sipRespBuffer);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(dummyEP.GetIPEndPoint()));
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
await uasTx.GotResponse(dummyEP, dummyEP, okResponse);
var dialogue = new SIPDialogue(uasTx);
Assert.NotNull(dialogue);
Assert.Equal(SIPURI.ParseSIPURI("sip:127.0.0.1:1234"), dialogue.RemoteTarget);
Assert.Equal(SIPEndPoint.ParseSIPEndPoint("udp:192.168.0.50:5080"), dialogue.RemoteSIPEndPoint);
logger.LogDebug("---------------------------------------------------");
}
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 SIPServerUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
UASInviteTransaction uasTransaction,
ISIPAccount sipAccount)
{
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_uasTransaction = uasTransaction;
m_sipAccount = sipAccount;
m_uasTransaction.UASInviteTransactionFailed += ClientTransactionFailed;
m_uasTransaction.UASInviteTransactionCancelled += UASTransactionCancelled;
}
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);
}
public SIPServerUserAgent(
SIPTransport sipTransport,
SIPEndPoint outboundProxy,
UASInviteTransaction uasTransaction,
ISIPAccount sipAccount)
{
m_sipTransport = sipTransport;
m_outboundProxy = outboundProxy;
m_uasTransaction = uasTransaction;
m_sipAccount = sipAccount;
m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
m_uasTransaction.UASInviteTransactionCancelled += UASTransactionCancelled;
m_uasTransaction.TransactionRemoved += new SIPTransactionRemovedDelegate(UASTransaction_TransactionRemoved);
}
/// <summary>
/// This method can be used to start the processing of a new incoming call request.
/// The user agent will is acting as a server for this operation and it can be considered
/// the opposite of the Call method. This is only the first step in answering an incoming
/// call. It can still be rejected or answered after this point.
/// </summary>
/// <param name="inviteRequest">The invite request representing the incoming call.</param>
/// <returns>An ID string that needs to be supplied when the call is answered or rejected
/// (used to manage multiple pending incoming calls).</returns>
public SIPServerUserAgent AcceptCall(SIPRequest inviteRequest)
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(m_transport, inviteRequest, m_outboundProxy);
SIPServerUserAgent uas = new SIPServerUserAgent(m_transport, m_outboundProxy, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.CallCancelled += (pendingUas) =>
{
CallEnded();
ServerCallCancelled?.Invoke(pendingUas);
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
return(uas);
}
public void AckRecognitionIIUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport sipTransport = new SIPTransport();
SIPTransactionEngine engine = sipTransport.m_transactionEngine; // Client side of the INVITE.
string inviteRequestStr =
"INVITE sip:[email protected] SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.1.2:5065;rport;branch=z9hG4bKFBB7EAC06934405182D13950BD51F001" + m_CRLF +
"From: SER Test X <sip:[email protected]:5065>;tag=196468136" + m_CRLF +
"To: <sip:[email protected]>" + m_CRLF +
"Contact: <sip:[email protected]:5065>" + m_CRLF +
"Call-ID: [email protected]" + m_CRLF +
"CSeq: 49429 INVITE" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"User-Agent: Dummy" + m_CRLF +
m_CRLF;
SIPRequest inviteRequest = SIPRequest.ParseSIPRequest(inviteRequestStr);
// Server has received the invite.
SIPEndPoint dummySIPEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 1234));
UASInviteTransaction serverTransaction = new UASInviteTransaction(sipTransport, inviteRequest, null, true);
engine.AddTransaction(serverTransaction);
string ackRequestStr =
"ACK sip:[email protected] SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.1.2:5065;rport;branch=z9hG4bKFBB7EAC06934405182D13950BD51F001" + m_CRLF +
"From: SER Test X <sip:[email protected]:5065>;tag=196468136" + m_CRLF +
"To: <sip:[email protected]>" + m_CRLF +
"Contact: <sip:[email protected]:5065>" + m_CRLF +
"Call-ID: [email protected]" + m_CRLF +
"CSeq: 49429 ACK" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: Dummy" + m_CRLF +
m_CRLF;
SIPRequest ackRequest = SIPRequest.ParseSIPRequest(ackRequestStr);
SIPTransaction matchingTransaction = engine.GetTransaction(ackRequest);
Assert.True(matchingTransaction.TransactionId == serverTransaction.TransactionId, "ACK transaction did not match INVITE transaction.");
}
public RTSPApp(
SIPMonitorLogDelegate statefulProxyLogEvent,
UASInviteTransaction clientTransaction,
string username)
{
m_statefulProxyLogEvent = statefulProxyLogEvent;
m_clientTransaction = clientTransaction;
//m_clientTransaction.TransactionCancelled += new SIPTransactionCancelledDelegate(CallCancelled);
m_clientUsername = username;
logger.Debug(m_clientTransaction.TransactionRequest.Body);
m_rtpPort = Convert.ToInt32(Regex.Match(m_clientTransaction.TransactionRequest.Body, @"m=audio (?<port>\d+)", RegexOptions.Singleline).Result("${port}"));
logger.Debug("RTP port=" + m_rtpPort);
}
public void AckRecognitionIIUnitTest()
{
SIPTransactionEngine engine = new SIPTransactionEngine(); // Client side of the INVITE.
string inviteRequestStr =
"INVITE sip:[email protected] SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.1.2:5065;rport;branch=z9hG4bKFBB7EAC06934405182D13950BD51F001" + m_CRLF +
"From: SER Test X <sip:[email protected]:5065>;tag=196468136" + m_CRLF +
"To: <sip:[email protected]>" + m_CRLF +
"Contact: <sip:[email protected]:5065>" + m_CRLF +
"Call-ID: [email protected]" + m_CRLF +
"CSeq: 49429 INVITE" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"User-Agent: Dummy" + m_CRLF +
m_CRLF;
SIPRequest inviteRequest = SIPRequest.ParseSIPRequest(inviteRequestStr);
// Server has received the invite.
SIPEndPoint dummySIPEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 1234));
UASInviteTransaction serverTransaction = new UASInviteTransaction(new SIPTransport(MockSIPDNSManager.Resolve, null), inviteRequest, dummySIPEndPoint, dummySIPEndPoint, null, IPAddress.Loopback, true);
engine.AddTransaction(serverTransaction);
//SIPResponse errorResponse = SIPTransport.GetResponse(inviteRequest.Header, SIPResponseStatusCodesEnum.Decline, "Unit Test", null, null);
string ackRequestStr =
"ACK sip:[email protected] SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.1.2:5065;rport;branch=z9hG4bKFBB7EAC06934405182D13950BD51F001" + m_CRLF +
"From: SER Test X <sip:[email protected]:5065>;tag=196468136" + m_CRLF +
"To: <sip:[email protected]>" + m_CRLF +
"Contact: <sip:[email protected]:5065>" + m_CRLF +
"Call-ID: [email protected]" + m_CRLF +
"CSeq: 49429 ACK" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: Dummy" + m_CRLF +
m_CRLF;
SIPRequest ackRequest = SIPRequest.ParseSIPRequest(ackRequestStr);
SIPTransaction matchingTransaction = engine.GetTransaction(ackRequest);
Assert.IsTrue(matchingTransaction.TransactionId == serverTransaction.TransactionId, "ACK transaction did not match INVITE transaction.");
}
/// <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>
/// This method can be used to start the processing of a new incoming call request.
/// The user agent will is acting as a server for this operation and it can be considered
/// the opposite of the Call method.
/// </summary>
/// <param name="uasInviteTx">The invite transaction representing the incoming call.</param>
/// <returns>True if the call is accepted, false otherwise.</returns>
public bool AcceptCall(UASInviteTransaction uasInviteTx)
{
SIPServerUserAgent uas = new SIPServerUserAgent(m_transport, m_outboundProxy, null, null, SIPCallDirection.In, null, null, null, uasInviteTx);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// TODO: Decide how to deal with multiple simultaneous calls.
if (m_uas != null)
{
uas.Reject(SIPResponseStatusCodesEnum.BusyHere, null, null);
return(false);
}
else
{
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
m_uas = uas;
return(true);
}
}
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($"B2BUA forwarding call to {dst.Uri}.");
b2bua.Call(dst);
}
}
}
/// <summary>
/// This function type is to allow B2B user agents to lookup the forwarding destination
/// for an accepted User Agent Server (UAS) call leg. The intent is that functions
/// can implement a form of a dialplan and pass to the B2BUA core.
/// </summary>
/// <param name="uas">A User Agent Server (UAS) transaction that has been accepted
/// for forwarding.</param>
/// <returns>A call descriptor for the User Agent Client (UAC) call leg that will
/// be bridged to the UAS leg.</returns>
public async Task <SIPCallDescriptor> Lookup(UASInviteTransaction uasTx, ISIPAccount from)
{
var dialplan = await LoadDialPlan();
//_logger.LogDebug($"Our dialplan last update {TrimMilliseconds(_dialplanLastUpdated).ToString("o")}, " +
// $"database last update {TrimMilliseconds(dialplan.LastUpdate).ToString("o")}.");
if (dialplan != null && TrimMilliseconds(DateTime.Parse(dialplan.LastUpdate)) > TrimMilliseconds(_dialplanLastUpdated))
{
_logger.LogInformation($"SIP DialPlan Manager loading updated dialplan.");
CompileDialPlan(dialplan.DialPlanScript, DateTime.Parse(dialplan.LastUpdate));
}
if (_dialPlanScriptRunner != null)
{
var result = await _dialPlanScriptRunner.Invoke(new SIPDialPlanGlobals { UasTx = uasTx, From = from });
return(result as SIPCallDescriptor);
}
else
{
return(null);
}
}
/// <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);
}
}
}
/// <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);
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
RtpAVSession rtpSession = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
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);
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)SDPMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
rtpSession = new RtpAVSession(
new AudioOptions {
AudioSource = AudioSourcesEnum.Music, SourceFile = executableDir + "/" + AUDIO_FILE_PCMU
}, null);
rtpSession.setRemoteDescription(new RTCSessionDescription {
type = RTCSdpType.offer, sdp = offerSdp
});
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
await sipTransport.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
rtpCts = new CancellationTokenSource();
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.CloseSession(null);
};
rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = await rtpSession.createAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
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();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
}
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($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
/// <summary>
/// Handler for when an in dialog request is received on an established call.
/// Typical types of request will be re-INVITES for things like putting a call on or
/// off hold and REFER requests for transfers. Some in dialog request types, such
/// as re-INVITES have specific events so they can be bubbled up to the
/// application to deal with.
/// </summary>
/// <param name="sipRequest">The in dialog request received.</param>
private async Task DialogRequestReceivedAsync(SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
logger.LogInformation($"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);
}
}
}
public async Task HandleInvalidSdpPortOnPlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
// This transport will act as the call receiver. It allows the test to provide a
// tailored response to an incoming call.
SIPTransport calleeTransport = new SIPTransport();
// This transport will be used by the SIPUserAgent being tested to place the call.
SIPTransport callerTransport = new SIPTransport();
RTPSession rtpSession = new RTPSession(false, false, false);
try
{
calleeTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
calleeTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
if (req.Method != SIPMethodsEnum.INVITE)
{
SIPResponse notAllowedResponse = SIPResponse.GetResponse(req, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await calleeTransport.SendResponseAsync(notAllowedResponse);
}
else
{
UASInviteTransaction uasTransaction = new UASInviteTransaction(calleeTransport, req, null);
var uas = new SIPServerUserAgent(calleeTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = @"
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp, null, SIPDialogueTransferModesEnum.NotAllowed);
}
};
SIPUserAgent userAgent = new SIPUserAgent(callerTransport, null);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
SIPURI dstUri = new SIPURI(SIPSchemesEnum.sip, calleeTransport.GetSIPChannels().First().ListeningSIPEndPoint);
var result = await userAgent.Call(dstUri.ToString(), null, null, rtpSession);
Assert.False(result);
}
finally
{
rtpSession?.Close("normal");
callerTransport?.Shutdown();
calleeTransport?.Shutdown();
}
}
public async Task HangupUserAgentUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport(false, MockSIPDNSManager.Resolve);
MockSIPChannel mockChannel = new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0));
transport.AddSIPChannel(mockChannel);
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 343" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF +
"Min-SE: 90" + m_CRLF +
"" + m_CRLF +
"v=0" + m_CRLF +
"o=- 3785527268 3785527269 IN IP4 192.168.11.50" + m_CRLF +
"s=pjmedia" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 4032 RTP/AVP 0 101" + m_CRLF +
"c=IN IP4 192.168.11.50" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
SIPServerUserAgent mockUas = new SIPServerUserAgent(transport, null, null, null, SIPCallDirection.In, null, null, null, uasTx);
await userAgent.Answer(mockUas, CreateMediaSession());
// Incremented Cseq and modified Via header from original request. Means the request is the same dialog but different tx.
string inviteReqStr2 = "BYE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3700" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3533 BYE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"";
mockChannel.FireMessageReceived(dummySipEndPoint, dummySipEndPoint, Encoding.UTF8.GetBytes(inviteReqStr2));
}
public void Call(SIPCallDescriptor sipCallDescriptor)
{
try
{
m_uacCallDescriptor = sipCallDescriptor;
SIPRequest uacInviteRequest = GetInviteRequest(m_uacCallDescriptor.Uri, sipCallDescriptor.GetFromHeader());
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;
}
//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);
}
catch (Exception excp)
{
logger.Error("Exception SIPB2BUserAgent Call. " + excp.Message);
}
}
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
}
//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
}
