示例#1
0
        static void Main(string[] args)
        {
            Console.WriteLine("SIPSorcery client user agent example.");
            Console.WriteLine("Press ctrl-c to exit.");

            // Plumbing code to facilitate a graceful exit.
            ManualResetEvent exitMre       = new ManualResetEvent(false);
            bool             isCallHungup  = false;
            bool             hasCallFailed = false;

            Log = AddConsoleLogger();

            SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);

            if (args != null && args.Length > 0)
            {
                if (!SIPURI.TryParse(args[0], out callUri))
                {
                    Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
                }
            }
            Log.LogInformation($"Call destination {callUri}.");

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

            EnableTraceLogs(sipTransport);

            var audioSession = new WindowsAudioEndPoint(new AudioEncoder());
            var rtpSession   = new VoIPMediaSession(audioSession.ToMediaEndPoints());

            var offerSDP = rtpSession.CreateOffer(null);

            // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
            var uac = new SIPClientUserAgent(sipTransport);

            uac.CallTrying  += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
            uac.CallRinging += async(uac, resp) =>
            {
                Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
                if (resp.Status == SIPResponseStatusCodesEnum.SessionProgress)
                {
                    await rtpSession.Start();
                }
            };
            uac.CallFailed += (uac, err, resp) =>
            {
                Log.LogWarning($"Call attempt to {uac.CallDescriptor.To} Failed: {err}");
                hasCallFailed = true;
            };
            uac.CallAnswered += async(iuac, resp) =>
            {
                if (resp.Status == SIPResponseStatusCodesEnum.Ok)
                {
                    Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");

                    var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body));
                    if (result == SetDescriptionResultEnum.OK)
                    {
                        await rtpSession.Start();
                    }
                    else
                    {
                        Log.LogWarning($"Failed to set remote description {result}.");
                        uac.Hangup();
                    }
                }
                else
                {
                    Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
                }
            };

            // The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
            sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
            {
                if (sipRequest.Method == SIPMethodsEnum.BYE)
                {
                    SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                    await sipTransport.SendResponseAsync(okResponse);

                    if (uac.IsUACAnswered)
                    {
                        Log.LogInformation("Call was hungup by remote server.");
                        isCallHungup = true;
                        exitMre.Set();
                    }
                }
            };

            // Start the thread that places the call.
            SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
                SIPConstants.SIP_DEFAULT_USERNAME,
                null,
                callUri.ToString(),
                SIPConstants.SIP_DEFAULT_FROMURI,
                callUri.CanonicalAddress,
                null, null, null,
                SIPCallDirection.Out,
                SDP.SDP_MIME_CONTENTTYPE,
                offerSDP.ToString(),
                null);

            uac.Call(callDescriptor, null);
            uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}");

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

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

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

            rtpSession.Close(null);

            if (!isCallHungup && uac != null)
            {
                if (uac.IsUACAnswered)
                {
                    Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
                    uac.Hangup();
                }
                else if (!hasCallFailed)
                {
                    Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
                    uac.Cancel();
                }

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

            if (sipTransport != null)
            {
                Log.LogInformation("Shutting down SIP transport...");
                sipTransport.Shutdown();
            }
        }
示例#2
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.");

            Log = AddConsoleLogger();

            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(SIPS_CERTIFICATE_PATH);

            // 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.
            VoIPMediaSession        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)
                    {
                        Log.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.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMU)))
                        {
                            Log.LogDebug($"Client offer contained PCMU audio codec.");
                            AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
                                AudioSource = AudioSourcesEnum.Music
                            });
                            rtpSession = new VoIPMediaSession(new MediaEndPoints {
                                AudioSource = extrasSource
                            });
                            rtpSession.AcceptRtpFromAny = true;

                            var setResult = rtpSession.SetRemoteDescription(SdpType.offer, offerSdp);

                            if (setResult != SetDescriptionResultEnum.OK)
                            {
                                // Didn't get a match on the codecs we support.
                                SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
                                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, uasTransaction, null);
                                uas.CallCancelled += (uasAgent) =>
                                {
                                    rtpCts?.Cancel();
                                    rtpSession.Close(null);
                                };
                                rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
                                uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
                                await Task.Delay(100);

                                uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
                                await Task.Delay(100);

                                var answerSdp = rtpSession.CreateAnswer(null);
                                uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);

                                await rtpSession.Start();
                            }
                        }
                    }
                    else if (sipRequest.Method == SIPMethodsEnum.BYE)
                    {
                        Log.LogInformation("Call hungup.");
                        SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
                        await sipTransport.SendResponseAsync(byeResponse);

                        uas?.Hangup(true);
                        rtpSession?.Close(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)
                {
                    Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
                }
            };

            ManualResetEvent exitMre = new ManualResetEvent(false);

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

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

                Hangup(uas).Wait();

                rtpSession?.Close(null);
                rtpCts?.Cancel();

                if (sipTransport != null)
                {
                    Log.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?.Close(null);
                            rtpCts?.Cancel();
                        }

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

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

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

            exitMre.WaitOne();
        }