static void Main() { Console.WriteLine("SIPSorcery call hold example."); Console.WriteLine("Press ctrl-c to exit."); // Plumbing code to facilitate a graceful exit. CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream. bool isCallHungup = false; bool hasCallFailed = false; AddConsoleLogger(); // Set up a default SIP transport. var sipTransport = new SIPTransport(); sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT))); //EnableTraceLogs(sipTransport); // Get the default speaker. var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice(); WaveInEvent waveInEvent = GetAudioInputDevice(); RTPMediaSession RtpMediaSession = null; // Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var userAgent = new SIPUserAgent(sipTransport, null); userAgent.ClientCallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); userAgent.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); userAgent.ClientCallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; exitCts.Cancel(); }; userAgent.ClientCallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); PlayRemoteMedia(RtpMediaSession, audioOutProvider); } else { Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); hasCallFailed = true; exitCts.Cancel(); } }; userAgent.OnCallHungup += () => { Log.LogInformation($"Call hungup by remote party."); exitCts.Cancel(); }; userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller."); sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) => { if (sipRequest.Header.From != null && sipRequest.Header.From.FromTag != null && sipRequest.Header.To != null && sipRequest.Header.To.ToTag != null) { // This is an in-dialog request that will be handled directly by a user agent instance. } else if (sipRequest.Method == SIPMethodsEnum.INVITE) { if (userAgent?.IsCallActive == true) { Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}."); // If we are already on a call return a busy response. UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null); SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null); uasTransaction.SendFinalResponse(busyResponse); } else { Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}."); var incomingCall = userAgent.AcceptCall(sipRequest); RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork); RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold."); RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold."); await userAgent.Answer(incomingCall, RtpMediaSession); PlayRemoteMedia(RtpMediaSession, audioOutProvider); waveInEvent.StartRecording(); Log.LogInformation($"Answered incoming call from {sipRequest.Header.From.FriendlyDescription()} at {remoteEndPoint}."); } } else { Log.LogDebug($"SIP {sipRequest.Method} request received but no processing has been set up for it, rejecting."); SIPResponse notAllowedResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null); await sipTransport.SendResponseAsync(notAllowedResponse); } }; // Wire up the RTP send session to the audio output device. uint rtpSendTimestamp = 0; waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) => { byte[] sample = new byte[args.Buffer.Length / 2]; int sampleIndex = 0; for (int index = 0; index < args.BytesRecorded; index += 2) { var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index)); sample[sampleIndex++] = ulawByte; } if (RtpMediaSession != null) { RtpMediaSession.SendAudioFrame(rtpSendTimestamp, sample); rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds); } }; // At this point the call has been initiated and everything will be handled in an event handler. Task.Run(async() => { try { while (!exitCts.Token.WaitHandle.WaitOne(0)) { var keyProps = Console.ReadKey(); if (keyProps.KeyChar == 'c') { if (!userAgent.IsCallActive) { RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, (int)SDPMediaFormatsEnum.PCMU, AddressFamily.InterNetwork); RtpMediaSession.RemotePutOnHold += () => Log.LogInformation("Remote call party has placed us on hold."); RtpMediaSession.RemoteTookOffHold += () => Log.LogInformation("Remote call party took us off hold."); var callDescriptor = GetCallDescriptor(DEFAULT_DESTINATION_SIP_URI); await userAgent.InitiateCall(callDescriptor, RtpMediaSession); } else { Log.LogWarning("There is already an active call."); } } else if (keyProps.KeyChar == 'h') { // Place call on/off hold. if (userAgent.IsCallActive) { if (RtpMediaSession.LocalOnHold) { Log.LogInformation("Taking the remote call party off hold."); RtpMediaSession.TakeOffHold(); } else { Log.LogInformation("Placing the remote call party on hold."); RtpMediaSession.PutOnHold(); } } else { Log.LogWarning("There is no active call to put on hold."); } } else if (keyProps.KeyChar == 't') { if (userAgent.IsCallActive) { var transferURI = SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI); bool result = await userAgent.BlindTransfer(transferURI, TimeSpan.FromSeconds(TRANSFER_TIMEOUT_SECONDS), exitCts.Token); if (result) { // If the transfer was accepted the original call will already have been hungup. // Wait a second for the transfer NOTIFY request to arrive. await Task.Delay(1000); exitCts.Cancel(); } else { Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed."); } } else { Log.LogWarning("There is no active call to transfer."); } } else if (keyProps.KeyChar == 'q') { // Quit application. exitCts.Cancel(); } } } catch (Exception excp) { SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}."); } }); // Ctrl-c will gracefully exit the call at any point. Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e) { e.Cancel = true; exitCts.Cancel(); }; // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed. exitCts.Token.WaitHandle.WaitOne(); #region Cleanup. Log.LogInformation("Exiting..."); RtpMediaSession?.Close(); waveInEvent?.StopRecording(); audioOutEvent?.Stop(); if (!isCallHungup && userAgent != null) { if (userAgent.IsCallActive) { Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}."); userAgent.Hangup(); } else if (!hasCallFailed) { Log.LogInformation($"Cancelling call to {userAgent?.CallDescriptor?.To}."); userAgent.Cancel(); } // Give the BYE or CANCEL request time to be transmitted. Log.LogInformation("Waiting 1s for call to clean up..."); Task.Delay(1000).Wait(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } #endregion }
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. 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($"Waiting for browser web socket connection to {_webSocketServer.Address}:{_webSocketServer.Port}..."); // Set up a default SIP transport. var sipTransport = new SIPTransport(); sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT))); //EnableTraceLogs(sipTransport); RTPMediaSession RtpMediaSession = null; // Create a SIP user agent to receive a call from a remote SIP client. // Wire up event handlers for the different stages of the call. var userAgent = new SIPUserAgent(sipTransport, null); // We're only answering SIP calls, not placing them. userAgent.OnCallHungup += () => { Log.LogInformation($"Call hungup by remote party."); exitCts.Cancel(); }; userAgent.ServerCallCancelled += (uas) => Log.LogInformation("Incoming call cancelled by caller."); sipTransport.SIPTransportRequestReceived += async(localEndPoint, remoteEndPoint, sipRequest) => { if (sipRequest.Header.From != null && sipRequest.Header.From.FromTag != null && sipRequest.Header.To != null && sipRequest.Header.To.ToTag != null) { // This is an in-dialog request that will be handled directly by a user agent instance. } else if (sipRequest.Method == SIPMethodsEnum.INVITE) { if (userAgent?.IsCallActive == true) { Log.LogWarning($"Busy response returned for incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}."); // If we are already on a call return a busy response. UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null); SIPResponse busyResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.BusyHere, null); uasTransaction.SendFinalResponse(busyResponse); } else { Log.LogInformation($"Incoming call request from {remoteEndPoint}: {sipRequest.StatusLine}."); var incomingCall = userAgent.AcceptCall(sipRequest); RtpMediaSession = new RTPMediaSession(SDPMediaTypesEnum.audio, new SDPMediaFormat(SDPMediaFormatsEnum.PCMU), AddressFamily.InterNetwork); await userAgent.Answer(incomingCall, RtpMediaSession); RtpMediaSession.OnRtpPacketReceived += (mediaType, rtpPacket) => OnMediaSampleReady?.Invoke(mediaType, rtpPacket.Header.Timestamp, rtpPacket.Payload); 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..."); RtpMediaSession?.Close(); if (userAgent != null) { if (userAgent.IsCallActive) { Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}."); userAgent.Hangup(); } // Give the BYE or CANCEL request time to be transmitted. Log.LogInformation("Waiting 1s for call to clean up..."); Task.Delay(1000).Wait(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } #endregion }