Exemple #1
0
        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);
                }
            }
        }
Exemple #2
0
            public void AckRecognitionUnitTest()
            {
                SIPTransport clientTransport = null;
                SIPTransport serverTransport = null;

                try
                {
                    SIPTransactionEngine clientEngine   = new SIPTransactionEngine();   // Client side of the INVITE.
                    SIPEndPoint          clientEndPoint =
                        new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 12013));
                    clientTransport =
                        new SIPTransport(MockSIPDNSManager.Resolve, clientEngine, new SIPUDPChannel(clientEndPoint.GetIPEndPoint()), false);
                    SetTransportTraceEvents(clientTransport);

                    SIPTransactionEngine serverEngine      = new SIPTransactionEngine(); // Server side of the INVITE.
                    UASInviteTransaction serverTransaction = null;
                    SIPEndPoint          serverEndPoint    = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 12014));
                    serverTransport =
                        new SIPTransport(MockSIPDNSManager.Resolve, serverEngine, new SIPUDPChannel(serverEndPoint.GetIPEndPoint()), false);
                    SetTransportTraceEvents(serverTransport);
                    serverTransport.SIPTransportRequestReceived += (localEndPoint, remoteEndPoint, sipRequest) =>
                    {
                        Console.WriteLine("Server Transport Request In: " + sipRequest.Method + ".");
                        serverTransaction =
                            serverTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localEndPoint, null);
                        SetTransactionTraceEvents(serverTransaction);
                        serverTransaction.GotRequest(localEndPoint, remoteEndPoint, sipRequest);
                    };

                    SIPURI     dummyURI      = SIPURI.ParseSIPURI("sip:dummy@" + serverEndPoint);
                    SIPRequest inviteRequest = GetDummyINVITERequest(dummyURI);
                    inviteRequest.LocalSIPEndPoint = clientTransport.GetDefaultTransportContact(SIPProtocolsEnum.udp);

                    // Send the invite to the server side.
                    UACInviteTransaction clientTransaction =
                        new UACInviteTransaction(clientTransport, inviteRequest, serverEndPoint, clientEndPoint, null);
                    SetTransactionTraceEvents(clientTransaction);
                    clientEngine.AddTransaction(clientTransaction);
                    clientTransaction.SendInviteRequest(serverEndPoint, inviteRequest);

                    Thread.Sleep(500);

                    Assert.IsTrue(clientTransaction.TransactionState == SIPTransactionStatesEnum.Completed, "Client transaction in incorrect state.");
                    Assert.IsTrue(serverTransaction.TransactionState == SIPTransactionStatesEnum.Confirmed, "Server transaction in incorrect state.");
                }
                finally
                {
                    if (clientTransport != null)
                    {
                        clientTransport.Shutdown();
                    }

                    if (serverTransport != null)
                    {
                        serverTransport.Shutdown();
                    }
                }
            }
        public void Call(SIPCallDescriptor sipCallDescriptor)
        {
            try
            {
                m_uacCallDescriptor = sipCallDescriptor;
                SIPRequest uacInviteRequest = GetInviteRequest(m_uacCallDescriptor.Uri, sipCallDescriptor);
                if (sipCallDescriptor.MangleResponseSDP && sipCallDescriptor.MangleIPAddress != null)
                {
                    uacInviteRequest.Header.ProxyReceivedFrom = sipCallDescriptor.MangleIPAddress.ToString();
                }

                uacInviteRequest.Body = sipCallDescriptor.Content;
                uacInviteRequest.Header.ContentType = sipCallDescriptor.ContentType;
                uacInviteRequest.LocalSIPEndPoint   = m_blackhole;
                uacInviteRequest.RemoteSIPEndPoint  = m_blackhole;

                // Now that we have a destination socket create a new UAC transaction for forwarded leg of the call.
                m_uacTransaction =
                    m_sipTransport.CreateUACTransaction(uacInviteRequest, m_blackhole, m_blackhole, null);
                if (m_uacTransaction.CDR != null)
                {
                    m_uacTransaction.CDR.Owner             = m_uacOwner;
                    m_uacTransaction.CDR.AdminMemberId     = m_uacAdminMemberId;
                    m_uacTransaction.CDR.DialPlanContextID = (m_uacCallDescriptor != null)
                        ? m_uacCallDescriptor.DialPlanContextID
                        : Guid.Empty;
                }

                //uacTransaction.UACInviteTransactionInformationResponseReceived += ServerInformationResponseReceived;
                //uacTransaction.UACInviteTransactionFinalResponseReceived += ServerFinalResponseReceived;
                //uacTransaction.UACInviteTransactionTimedOut += ServerTimedOut;
                //uacTransaction.TransactionTraceMessage += TransactionTraceMessage;

                m_uacTransaction.SendInviteRequest(m_blackhole, m_uacTransaction.TransactionRequest);

                SIPRequest uasInviteRequest = uacInviteRequest.Copy();
                uasInviteRequest.LocalSIPEndPoint  = m_blackhole;
                uasInviteRequest.RemoteSIPEndPoint = m_blackhole;
                uasInviteRequest.Header.Vias.TopViaHeader.Branch = CallProperties.CreateBranchId();
                m_uasTransaction =
                    m_sipTransport.CreateUASTransaction(uasInviteRequest, m_blackhole, m_blackhole, null);

                SetOwner(sipCallDescriptor.ToSIPAccount.Owner, sipCallDescriptor.ToSIPAccount.AdminMemberId);
                //m_uasTransaction.TransactionTraceMessage += TransactionTraceMessage;
                //m_uasTransaction.UASInviteTransactionTimedOut += ClientTimedOut;
                //m_uasTransaction.UASInviteTransactionCancelled += (t) => { };

                QueueNewCall_External(this);

                CallTrying.Invoke(this, null);
            }
            catch (Exception excp)
            {
                logger.Error("Exception SIPB2BUserAgent Call. " + excp.Message);
            }
        }
Exemple #4
0
        /// <summary>
        /// 发送可靠的sip请求消息
        /// </summary>
        /// <param name="remoteEP">远程终结点</param>
        /// <param name="request">sip请求消息</param>
        public void SendReliableRequest(SIPEndPoint remoteEP, SIPRequest request)
        {
            if (_serviceState == ServiceStatus.Wait)
            {
                OnSIPServiceChange(remoteEP.ToHost(), ServiceStatus.Wait);
                return;
            }

            SIPTransaction trans = Transport.CreateUASTransaction(request, remoteEP, LocalEP, null);

            trans.TransactionStateChanged += trans_TransactionStateChanged;
            trans.SendReliableRequest();
        }
        private DialPlanLineContext GetDummyDialPlanContext(string testDialPlan, string dst)
        {
            SIPDialPlan          dialPlan          = new SIPDialPlan(null, null, null, testDialPlan, SIPDialPlanScriptTypesEnum.Asterisk);
            SIPTransactionEngine transactionEngine = new SIPTransactionEngine();
            SIPTransport         sipTransport      = new SIPTransport(MockSIPDNSManager.Resolve, transactionEngine);
            SIPURI               dummyURI          = SIPURI.ParseSIPURI(dst);
            SIPRequest           inviteRequest     = GetDummyINVITERequest(dummyURI);
            SIPEndPoint          dummyEndPoint     = SIPEndPoint.ParseSIPEndPoint("udp:0.0.0.0:5060");
            UASInviteTransaction uasTransaction    = sipTransport.CreateUASTransaction(inviteRequest, dummyEndPoint, dummyEndPoint, null);
            SIPServerUserAgent   uas             = new SIPServerUserAgent(sipTransport, null, "test", "sipsorcery.com", SIPCallDirection.In, null, null, null, uasTransaction);
            DialPlanLineContext  dialPlanContext = new DialPlanLineContext(null, null, null, null, uas, dialPlan, null, null, null, null);

            return(dialPlanContext);
        }
Exemple #6
0
        /// <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);
            }
        }
Exemple #7
0
        static void Main(string[] args)
        {
            Console.WriteLine("SIPSorcery user agent server example.");
            Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");

            EnableConsoleLogger();

            IPAddress listenAddress     = IPAddress.Any;
            IPAddress listenIPv6Address = IPAddress.IPv6Any;

            if (args != null && args.Length > 0)
            {
                if (!IPAddress.TryParse(args[0], out var customListenAddress))
                {
                    Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
                    listenAddress = IPAddress.Any;
                }
                else
                {
                    if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
                    {
                        listenAddress = customListenAddress;
                    }
                    if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
                    {
                        listenIPv6Address = customListenAddress;
                    }
                }
            }

            // Set up a default SIP transport.
            var sipTransport = new SIPTransport();

            var localhostCertificate = new X509Certificate2("localhost.pfx");

            // IPv4 channels.
            sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
            sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
            sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
            sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
            sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));

            // IPv6 channels.
            sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
            sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
            sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
            sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
            sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));

            EnableTraceLogs(sipTransport);

            // To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
            // acts as a singleton
            SIPServerUserAgent      uas    = null;
            CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
            Socket rtpSocket     = null;
            Socket controlSocket = null;

            // Because this is a server user agent the SIP transport must start listening for client user agents.
            sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
            {
                try
                {
                    if (sipRequest.Method == SIPMethodsEnum.INVITE)
                    {
                        SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");

                        // Check there's a codec we support in the INVITE offer.
                        var        offerSdp       = SDP.ParseSDPDescription(sipRequest.Body);
                        IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
                        RTPSession rtpSession     = null;
                        string     audioFile      = null;

                        if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.G722)))
                        {
                            Log.LogDebug($"Using G722 RTP media type and audio file {AUDIO_FILE_G722}.");
                            rtpSession = new RTPSession((int)RTPPayloadTypesEnum.G722, null, null);
                            audioFile  = AUDIO_FILE_G722;
                        }
                        else if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
                        {
                            Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
                            rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
                            audioFile  = AUDIO_FILE_PCMU;
                        }

                        if (rtpSession == null)
                        {
                            // Didn't get a match on the codecs we support.
                            SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
                            sipTransport.SendResponse(noMatchingCodecResponse);
                        }
                        else
                        {
                            // If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
                            // means this example can be kept simpler.
                            if (uas?.IsHungup == false)
                            {
                                uas?.Hangup(false);
                            }
                            rtpCts?.Cancel();

                            UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
                            uas    = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
                            rtpCts = new CancellationTokenSource();

                            uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
                            uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);

                            // Initialise an RTP session to receive the RTP packets from the remote SIP server.
                            NetServices.CreateRtpSocket(dstRtpEndPoint.AddressFamily == AddressFamily.InterNetworkV6 ? IPAddress.IPv6Any : IPAddress.Any, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);

                            // The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
                            IPAddress  rtpAddress  = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
                            IPEndPoint rtpEndPoint = new IPEndPoint(rtpAddress, (rtpSocket.LocalEndPoint as IPEndPoint).Port);

                            var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, audioFile, rtpCts))
                                          .ContinueWith(_ =>
                            {
                                if (uas?.IsHungup == false)
                                {
                                    uas?.Hangup(false);
                                }
                            });

                            uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
                        }
                    }
                    else if (sipRequest.Method == SIPMethodsEnum.BYE)
                    {
                        SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
                        SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
                        SIPResponse             byeResponse    = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                        byeTransaction.SendFinalResponse(byeResponse);
                        uas?.Hangup(true);
                        rtpCts?.Cancel();
                        rtpSocket?.Close();
                        controlSocket?.Close();
                    }
                    else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
                    {
                        SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
                        sipTransport.SendResponse(notAllowededResponse);
                    }
                    else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
                    {
                        SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                        sipTransport.SendResponse(optionsResponse);
                    }
                }
                catch (Exception reqExcp)
                {
                    SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
                }
            };

            ManualResetEvent exitMre = new ManualResetEvent(false);

            Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
            {
                e.Cancel = true;

                SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");

                Hangup(uas).Wait();

                rtpCts?.Cancel();
                rtpSocket?.Close();
                controlSocket?.Close();

                if (sipTransport != null)
                {
                    SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
                    sipTransport.Shutdown();
                }

                exitMre.Set();
            };

            Task.Run(() =>
            {
                try
                {
                    while (!exitMre.WaitOne(0))
                    {
                        var keyProps = Console.ReadKey();
                        if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
                        {
                            Console.WriteLine();
                            Console.WriteLine("Hangup requested by user...");

                            Hangup(uas).Wait();

                            rtpCts?.Cancel();
                            rtpSocket?.Close();
                            controlSocket?.Close();
                        }

                        if (keyProps.KeyChar == 'q')
                        {
                            SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");

                            if (sipTransport != null)
                            {
                                SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
                                sipTransport.Shutdown();
                            }

                            exitMre.Set();
                        }
                    }
                }
                catch (Exception excp)
                {
                    SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
                }
            });

            exitMre.WaitOne();
        }
        private static readonly int RTP_REPORTING_PERIOD_SECONDS = 5;       // Period at which to write RTP stats.

        static void Main()
        {
            Console.WriteLine("SIPSorcery client user agent server example.");
            Console.WriteLine("Press ctrl-c to exit.");

            // Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
            var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
            var loggerConfig  = new LoggerConfiguration()
                                .Enrich.FromLogContext()
                                .MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
                                .WriteTo.Console()
                                .CreateLogger();

            loggerFactory.AddSerilog(loggerConfig);
            SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;

            // Set up a default SIP transport.
            IPAddress defaultAddr  = LocalIPConfig.GetDefaultIPv4Address();
            var       sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new SIPTransactionEngine());
            int       port         = FreePort.FindNextAvailableUDPPort(SIPConstants.DEFAULT_SIP_PORT);
            var       sipChannel   = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));

            sipTransport.AddSIPChannel(sipChannel);

            // To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
            // acts as a singleton
            SIPServerUserAgent      uas    = null;
            CancellationTokenSource uasCts = null;

            // Because this is a server user agent the SIP transport must start listening for client user agents.
            sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
            {
                if (sipRequest.Method == SIPMethodsEnum.INVITE)
                {
                    SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");

                    // If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
                    // means this example can be kept a little it simpler.
                    uas?.Hangup();

                    UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
                    uas    = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
                    uasCts = new CancellationTokenSource();

                    uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
                    uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);

                    // Initialise an RTP session to receive the RTP packets from the remote SIP server.
                    Socket rtpSocket     = null;
                    Socket controlSocket = null;
                    NetServices.CreateRtpSocket(defaultAddr, 49000, 49100, false, out rtpSocket, out controlSocket);

                    IPEndPoint rtpEndPoint    = rtpSocket.LocalEndPoint as IPEndPoint;
                    IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
                    var        rtpSession     = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);

                    var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, uasCts))
                                  .ContinueWith(_ => { if (uas?.IsHungup == false)
                                                       {
                                                           uas?.Hangup();
                                                       }
                                                });

                    uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
                }
                else if (sipRequest.Method == SIPMethodsEnum.BYE)
                {
                    SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
                    SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
                    SIPResponse             byeResponse    = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                    byeTransaction.SendFinalResponse(byeResponse);
                    uas?.Hangup();
                    uasCts?.Cancel();
                }
            };

            Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
            {
                e.Cancel = true;

                SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
                if (uas?.IsHungup == false)
                {
                    uas?.Hangup();
                }
                uasCts?.Cancel();

                if (sipTransport != null)
                {
                    SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
                    sipTransport.Shutdown();
                }
            };
        }
Exemple #9
0
        private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
        {
            if (sipRequest.Method == SIPMethodsEnum.BYE)
            {
                var rtpJob = (from job in m_rtpJobs.Values where job.UAS.CallRequest.Header.CallId == sipRequest.Header.CallId select job).FirstOrDefault();

                if (rtpJob != null)
                {
                    rtpJob.Stop();
                    // Call has been hungup by remote end.
                    Console.WriteLine("Call hungup by client: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
                    Publish(rtpJob.QueueName, "BYE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
                    //Console.WriteLine("Request Received " + localSIPEndPoint + "<-" + remoteEndPoint + "\n" + sipRequest.ToString());
                    //m_uas.SIPDialogue.Hangup(m_sipTransport, null);
                    SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                    m_sipTransport.SendResponse(okResponse);
                }
                else
                {
                    Console.WriteLine("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".\n");
                    SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
                    m_sipTransport.SendResponse(noCallLegResponse);
                }
            }
            else if (sipRequest.Method == SIPMethodsEnum.INVITE)
            {
                Console.WriteLine("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
                Publish(sipRequest.URI.User, "INVITE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");

                Console.WriteLine(sipRequest.Body);

                SIPPacketMangler.MangleSIPRequest(SIPMonitorServerTypesEnum.Unknown, sipRequest, null, LogTraceMessage);

                UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
                var uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, LogTraceMessage, uasTransaction);
                uas.CallCancelled += UASCallCancelled;

                RTPDiagnosticsJob rtpJob = new RTPDiagnosticsJob(m_rtpListenIPAddress, m_publicIPAddress, uas, sipRequest);

                string sdpAddress = SDP.GetSDPRTPEndPoint(sipRequest.Body).Address.ToString();

                // Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.
                IPEndPoint expectedRTPEndPoint = new IPEndPoint(rtpJob.RemoteRTPEndPoint.Address, rtpJob.RemoteRTPEndPoint.Port);
                if (IPSocket.IsPrivateAddress(rtpJob.RemoteRTPEndPoint.Address.ToString()))
                {
                    expectedRTPEndPoint.Address = remoteEndPoint.Address;
                }

                Publish(sipRequest.URI.User, "Advertised RTP remote socket " + rtpJob.RemoteRTPEndPoint + ", expecting from " + expectedRTPEndPoint + ".");
                m_rtpJobs.Add(rtpJob.RTPListenEndPoint.Port, rtpJob);

                //ThreadPool.QueueUserWorkItem(delegate { StartRTPListener(rtpJob); });

                Console.WriteLine(rtpJob.LocalSDP.ToString());

                uas.Answer("application/sdp", rtpJob.LocalSDP.ToString(), CallProperties.CreateNewTag(), null, SIPDialogueTransferModesEnum.NotAllowed);

                var hangupTimer = new Timer(delegate
                {
                    if (!rtpJob.StopJob)
                    {
                        if (uas != null && uas.SIPDialogue != null)
                        {
                            if (rtpJob.RTPPacketReceived && !rtpJob.ErrorOnRTPSend)
                            {
                                Publish(sipRequest.URI.User, "Test completed. There were no RTP send or receive errors.");
                            }
                            else if (!rtpJob.RTPPacketReceived)
                            {
                                Publish(sipRequest.URI.User, "Test completed. An error was identified, no RTP packets were received.");
                            }
                            else
                            {
                                Publish(sipRequest.URI.User, "Test completed. An error was identified, there was a problem when attempting to send an RTP packet.");
                            }
                            rtpJob.Stop();
                            uas.SIPDialogue.Hangup(m_sipTransport, null);
                        }
                    }
                }, null, HANGUP_TIMEOUT, Timeout.Infinite);
            }
            else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
            {
                UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));

                if (inviteTransaction != null)
                {
                    Console.WriteLine("Matching CANCEL request received " + sipRequest.URI.ToString() + ".\n");
                    Publish(sipRequest.URI.User, "CANCEL request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
                    SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
                    cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
                }
                else
                {
                    Console.WriteLine("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".\n");
                    SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
                    m_sipTransport.SendResponse(noCallLegResponse);
                }
            }
            else
            {
                Console.WriteLine("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.\n");
                Publish(sipRequest.URI.User, sipRequest.Method + " request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
                SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
                m_sipTransport.SendResponse(notAllowedResponse);
            }
        }
Exemple #10
0
        static void Main()
        {
            Console.WriteLine("SIPSorcery client user agent example.");
            Console.WriteLine("Press ctrl-c to exit.");

            CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.

            AddConsoleLogger();

            // Set up a default SIP transport.
            var sipTransport = new SIPTransport();

            sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));

            // Un/comment this line to see/hide each SIP message sent and received.
            EnableTraceLogs(sipTransport);

            // To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
            // acts as a singleton
            SIPUserAgent            userAgent = new SIPUserAgent(sipTransport, null);
            CancellationTokenSource rtpCts    = null; // Cancellation token to stop the RTP stream.
            Socket rtpSocket     = null;
            Socket controlSocket = null;

            // Because this is a server user agent the SIP transport must start listening for client user agents.
            sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
            {
                try
                {
                    if (sipRequest.Header.From != null &&
                        sipRequest.Header.From.FromTag != null &&
                        sipRequest.Header.To != null &&
                        sipRequest.Header.To.ToTag != null)
                    {
                        userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
                    }
                    if (sipRequest.Method == SIPMethodsEnum.INVITE)
                    {
                        SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");

                        // Check there's a codec we support in the INVITE offer.
                        var        offerSdp       = SDP.ParseSDPDescription(sipRequest.Body);
                        IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
                        RTPSession rtpSession     = null;
                        string     audioFile      = null;

                        if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
                        {
                            Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
                            rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
                            audioFile  = AUDIO_FILE_PCMU;
                        }

                        if (rtpSession == null)
                        {
                            // Didn't get a match on the codecs we support.
                            SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
                            sipTransport.SendResponse(noMatchingCodecResponse);
                        }
                        else
                        {
                            // If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
                            // means this example can be kept simpler.
                            if (userAgent?.IsAnswered == true)
                            {
                                userAgent?.Hangup();
                            }
                            rtpCts?.Cancel();

                            UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
                            if (userAgent.AcceptCall(uasTransaction))
                            {
                                rtpCts = new CancellationTokenSource();

                                // The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
                                IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
                                // Initialise an RTP session to receive the RTP packets from the remote SIP server.
                                NetServices.CreateRtpSocket(rtpAddress, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);

                                var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
                                var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
                                rtpSendSession.DestinationEndPoint           = dstRtpEndPoint;
                                rtpRecvSession.OnReceiveFromEndPointChanged += (oldEP, newEP) =>
                                {
                                    Log.LogDebug($"RTP destination end point changed from {oldEP} to {newEP}.");
                                    rtpSendSession.DestinationEndPoint = newEP;
                                };

                                Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
                                Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));

                                userAgent.Answer(GetSDP(rtpSocket.LocalEndPoint as IPEndPoint));
                            }
                        }
                    }
                    else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
                    {
                        SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
                        sipTransport.SendResponse(notAllowededResponse);
                    }
                    else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
                    {
                        SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                        sipTransport.SendResponse(optionsResponse);
                    }
                }
                catch (Exception reqExcp)
                {
                    SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
                }
            };

            // Ctrl-c will gracefully exit the call at any point.
            Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
            {
                e.Cancel = true;
                exitCts.Cancel();
                rtpCts?.Cancel();
            };

            // At this point the call has been initiated and everything will be handled in an event handler.
            Task.Run(async() =>
            {
                try
                {
                    while (!exitCts.Token.WaitHandle.WaitOne(0))
                    {
                        var keyProps = Console.ReadKey();
                        if (keyProps.KeyChar == 't')
                        {
                            // Initiate a transfer.
                            bool transferResult = await userAgent.Transfer(SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI), new TimeSpan(0, 0, TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
                            if (transferResult)
                            {
                                // If the transfer was accepted the original call will already have been hungup.
                                userAgent = null;
                                exitCts.Cancel();
                            }
                            else
                            {
                                Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
                            }
                        }
                        else if (keyProps.KeyChar == 'q')
                        {
                            // Quit application.
                            exitCts.Cancel();
                        }
                    }
                }
                catch (Exception excp)
                {
                    Log.LogError($"Exception Key Press listener. {excp.Message}.");
                }
            });

            // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
            exitCts.Token.WaitHandle.WaitOne();

            Log.LogInformation("Exiting...");

            rtpSocket?.Close();
            controlSocket?.Close();

            if (userAgent != null)
            {
                if (userAgent.IsAnswered)
                {
                    Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
                    userAgent.Hangup();
                }

                // Give the final request time to be transmitted.
                Log.LogInformation("Waiting 1s for call to clean up...");
                Task.Delay(1000).Wait();
            }

            SIPSorcery.Net.DNSManager.Stop();

            if (sipTransport != null)
            {
                Log.LogInformation("Shutting down SIP transport...");
                sipTransport.Shutdown();
            }
        }
Exemple #11
0
        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();
                }
            };
        }
Exemple #12
0
        /// <summary>
        /// Handler for when an in dialog request is received on an established call.
        /// Typical types of request will be re-INVITES for things like putting a call on or
        /// off hold and REFER requests for transfers. Some in dialog request types, such
        /// as re-INVITES have specific events so they can be bubbled up to the
        /// application to deal with.
        /// </summary>
        /// <param name="request">The in dialog request received.</param>
        public async Task InDialogRequestReceivedAsync(SIPRequest sipRequest)
        {
            // Make sure the request matches our dialog and is not a stray.
            // A dialog request should match on to tag, from tag and call ID. We'll be more
            // accepting just in case the sender got the tags wrong.
            if (Dialogue == null || sipRequest.Header.CallId != Dialogue.CallId)
            {
                var noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
                var sendResult        = await SendResponse(noCallLegResponse);

                if (sendResult != SocketError.Success)
                {
                    logger.LogWarning($"SIPUserAgent send response failed in InCallRequestReceivedAsync with {sendResult}.");
                }
            }
            else
            {
                if (sipRequest.Method == SIPMethodsEnum.BYE)
                {
                    logger.LogDebug($"Matching dialogue found for {sipRequest.StatusLine}.");

                    SIPNonInviteTransaction byeTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
                    SIPResponse             byeResponse    = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                    byeTransaction.SendFinalResponse(byeResponse);

                    CallHungup?.Invoke();

                    m_uac = null;
                    m_uas = null;
                }
                else if (sipRequest.Method == SIPMethodsEnum.INVITE)
                {
                    logger.LogDebug($"Re-INVITE request received {sipRequest.StatusLine}.");

                    UASInviteTransaction reInviteTransaction = m_transport.CreateUASTransaction(sipRequest, m_outboundProxy);

                    if (OnReinviteRequest == null)
                    {
                        // The application isn't prepared to accept re-INVITE requests. We'll reject as gently as we can to try and not lose the call.
                        SIPResponse notAcceptableResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptable, null);
                        reInviteTransaction.SendFinalResponse(notAcceptableResponse);
                    }
                    else
                    {
                        SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
                        reInviteTransaction.SendProvisionalResponse(tryingResponse);
                        OnReinviteRequest(reInviteTransaction);
                    }
                }
                else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
                {
                    //Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "OPTIONS request for established dialogue " + dialogue.DialogueName + ".", dialogue.Owner));
                    SIPNonInviteTransaction optionsTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
                    SIPResponse             okResponse         = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                    okResponse.Body = Dialogue.RemoteSDP;
                    okResponse.Header.ContentLength = okResponse.Body.Length;
                    okResponse.Header.ContentType   = m_sdpContentType;
                    optionsTransaction.SendFinalResponse(okResponse);
                }
                else if (sipRequest.Method == SIPMethodsEnum.MESSAGE)
                {
                    //Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "MESSAGE for call " + sipRequest.URI.ToString() + ": " + sipRequest.Body + ".", dialogue.Owner));
                    SIPNonInviteTransaction messageTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);
                    SIPResponse             okResponse         = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                    messageTransaction.SendFinalResponse(okResponse);
                }
                else if (sipRequest.Method == SIPMethodsEnum.REFER)
                {
                    //Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "REFER received on dialogue " + dialogue.DialogueName + ", transfer mode is " + dialogue.TransferMode + ".", dialogue.Owner));

                    SIPNonInviteTransaction referTransaction = m_transport.CreateNonInviteTransaction(sipRequest, m_outboundProxy);

                    if (sipRequest.Header.ReferTo.IsNullOrBlank())
                    {
                        // A REFER request must have a Refer-To header.
                        //Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Bad REFER request, no Refer-To header.", dialogue.Owner));
                        SIPResponse invalidResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BadRequest, "Missing mandatory Refer-To header");
                        referTransaction.SendFinalResponse(invalidResponse);
                    }
                    else
                    {
                        //TODO: Add handling logic for indialog REFER requests.
                    }
                }
            }
        }
        private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
        {
            var endpoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);

            if (sipRequest.Method == SIPMethodsEnum.INVITE)
            {
                if (transaction != null)
                {
                    return;
                }

                logger.DebugFormat("{0} Incoming call from {1}", prefix, sipRequest.Header.From.FromURI.User);

                transaction = transport.CreateUASTransaction(sipRequest, remoteEndPoint, endpoint, null);
                agent       = new SIPServerUserAgent(
                    transport,
                    null,
                    sipRequest.Header.From.FromURI.User,
                    null,
                    SIPCallDirection.In,
                    null,
                    null,
                    null,
                    transaction);

                agent.CallCancelled       += Agent_CallCancelled;
                agent.TransactionComplete += Agent_TransactionComplete;

                agent.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
                agent.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);

                var answer  = SDP.ParseSDPDescription(agent.CallRequest.Body);
                var address = IPAddress.Parse(answer.Connection.ConnectionAddress);
                var port    = answer.Media.FirstOrDefault(m => m.Media == SDPMediaTypesEnum.audio).Port;
                var random  = Crypto.GetRandomInt(5).ToString();
                var sdp     = new SDP
                {
                    Version     = 2,
                    Username    = "******",
                    SessionId   = random,
                    Address     = localIPEndPoint.Address.ToString(),
                    SessionName = "redfox_" + random,
                    Timing      = "0 0",
                    Connection  = new SDPConnectionInformation(publicIPAddress.ToString())
                };

                rtpChannel = new RTPChannel
                {
                    DontTimeout    = true,
                    RemoteEndPoint = new IPEndPoint(address, port)
                };

                rtpChannel.SetFrameType(FrameTypesEnum.Audio);
                // TODO Fix hardcoded ports
                rtpChannel.ReservePorts(15000, 15090);
                rtpChannel.OnFrameReady += Channel_OnFrameReady;
                rtpChannel.Start();

                // Send some setup parameters to punch a hole in the firewall/router
                rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });

                rtpChannel.OnControlDataReceived       += (b) => { logger.Debug($"{prefix} Control Data Received; {b.Length} bytes"); };
                rtpChannel.OnControlSocketDisconnected += () => { logger.Debug($"{prefix} Control Socket Disconnected"); };

                var announcement = new SDPMediaAnnouncement
                {
                    Media        = SDPMediaTypesEnum.audio,
                    MediaFormats = new List <SDPMediaFormat>()
                    {
                        new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
                    },
                    Port = rtpChannel.RTPPort
                };

                sdp.Media.Add(announcement);

                SetState(State.Listening, sipRequest.Header.From.FromURI.User);

                agent.Progress(SIPResponseStatusCodesEnum.Accepted, null, null, null, null);
                agent.Answer(SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);

                SetState(State.Busy, "");
                return;
            }
            if (sipRequest.Method == SIPMethodsEnum.BYE)
            {
                if (State != State.Busy)
                {
                    return;
                }

                logger.DebugFormat("{0} Hangup from {1}", prefix, sipRequest.Header.From.FromURI.User);

                var noninvite = transport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, endpoint, null);
                var response  = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);

                noninvite.SendFinalResponse(response);

                SetState(State.Finished, Endpoint);

                rtpChannel.OnFrameReady -= Channel_OnFrameReady;
                rtpChannel.Close();

                agent.TransactionComplete -= Agent_TransactionComplete;
                agent.CallCancelled       -= Agent_CallCancelled;
                agent       = null;
                transaction = null;

                SetState(State.Ready, Endpoint);

                return;
            }
            if (sipRequest.Method == SIPMethodsEnum.ACK)
            {
            }
            if (sipRequest.Method == SIPMethodsEnum.CANCEL)
            {
            }
        }