private static int INPUT_SAMPLE_PERIOD_MILLISECONDS = 20; // This sets the frequency of the RTP packets. 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; 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); // Get the IP address the RTP will be sent from. While we can listen on IPAddress.Any | IPv6Any // we can't put 0.0.0.0 or [::0] in the SDP or the callee will ignore us. var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false); Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}."); var dstAddress = lookupResult.GetSIPEndPoint().Address; IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress); // Initialise an RTP session to receive the RTP packets from the remote SIP server. var rtpSession = new RTPMediaSession((int)SDPMediaFormatsEnum.PCMU, localIPAddress.AddressFamily); var offerSDP = rtpSession.GetSDP(localIPAddress); // Get the audio input device. WaveInEvent waveInEvent = GetAudioInputDevice(); // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var uac = new SIPClientUserAgent(sipTransport); uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); // Only set the remote RTP end point if there hasn't already been a packet received on it. if (rtpSession.DestinationEndPoint == null) { rtpSession.DestinationEndPoint = SDP.GetSDPRTPEndPoint(resp.Body); Log.LogDebug($"Remote RTP socket {rtpSession.DestinationEndPoint}."); } rtpSession.SetRemoteSDP(SDP.ParseSDPDescription(resp.Body)); waveInEvent.StartRecording(); } 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(); } } }; // Wire up the RTP receive session to the audio output device. var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice(); rtpSession.OnReceivedSampleReady += (sample) => { for (int index = 0; index < sample.Length; index++) { short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]); byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) }; audioOutProvider.AddSamples(pcmSample, 0, 2); } }; // Wire up the RTP send session to the audio input device. uint rtpSendTimestamp = 0; waveInEvent.DataAvailable += (object sender, WaveInEventArgs args) => { byte[] sample = new byte[args.Buffer.Length / 2]; int sampleIndex = 0; for (int index = 0; index < args.BytesRecorded; index += 2) { var ulawByte = NAudio.Codecs.MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(args.Buffer, index)); sample[sampleIndex++] = ulawByte; } if (rtpSession.DestinationEndPoint != null) { rtpSession.SendAudioFrame(rtpSendTimestamp, sample); rtpSendTimestamp += (uint)(8000 / waveInEvent.BufferMilliseconds); } }; // 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); 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..."); waveInEvent?.StopRecording(); audioOutEvent?.Stop(); rtpSession.CloseSession(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(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } }
public string ToMangledContactString() { int secondsRemaining = Convert.ToInt32(ExpiryTime.Subtract(DateTime.UtcNow).TotalSeconds % Int32.MaxValue); return("<" + m_mangledContactURI.ToString() + ">;" + SIPContactHeader.EXPIRES_PARAMETER_KEY + "=" + secondsRemaining); }
public async Task HandleInvalidSdpPortOnPlaceCallUnitTest() { logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name); logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name); // This transport will act as the call receiver. It allows the test to provide a // tailored response to an incoming call. SIPTransport calleeTransport = new SIPTransport(); // This transport will be used by the SIPUserAgent being tested to place the call. SIPTransport callerTransport = new SIPTransport(); RTPSession rtpSession = new RTPSession(false, false, false); try { calleeTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0)); calleeTransport.SIPTransportRequestReceived += async(lep, rep, req) => { if (req.Method != SIPMethodsEnum.INVITE) { SIPResponse notAllowedResponse = SIPResponse.GetResponse(req, SIPResponseStatusCodesEnum.MethodNotAllowed, null); await calleeTransport.SendResponseAsync(notAllowedResponse); } else { UASInviteTransaction uasTransaction = new UASInviteTransaction(calleeTransport, req, null); var uas = new SIPServerUserAgent(calleeTransport, null, null, null, SIPCallDirection.In, null, null, uasTransaction); uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null); uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null); var answerSdp = @" v=0 o=- 1838015445 0 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 79762 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv"; uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp, null, SIPDialogueTransferModesEnum.NotAllowed); } }; SIPUserAgent userAgent = new SIPUserAgent(callerTransport, null); MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.PCMU) }); rtpSession.addTrack(audioTrack); SIPURI dstUri = new SIPURI(SIPSchemesEnum.sip, calleeTransport.GetSIPChannels().First().ListeningSIPEndPoint); var result = await userAgent.Call(dstUri.ToString(), null, null, rtpSession); Assert.False(result); } finally { rtpSession?.Close("normal"); callerTransport?.Shutdown(); calleeTransport?.Shutdown(); } }
/// <summary> /// Places an outgoing SIP call. /// </summary> /// <param name="destination">The SIP URI to place a call to. The destination can be a full SIP URI in which case the all will /// be placed anonymously directly to that URI. Alternatively it can be just the user portion of a URI in which case it will /// be sent to the configured SIP server.</param> public async Task Call(string destination) { // Determine if this is a direct anonymous call or whether it should be placed using the pre-configured SIP server account. SIPURI callURI = null; string sipUsername = null; string sipPassword = null; string fromHeader = null; if (destination.Contains("@") || m_sipServer == null) { // Anonymous call direct to SIP server specified in the URI. callURI = SIPURI.ParseSIPURIRelaxed(destination); fromHeader = (new SIPFromHeader(m_sipFromName, SIPURI.ParseSIPURI(SIPFromHeader.DEFAULT_FROM_URI), null)).ToString(); } else { // This call will use the pre-configured SIP account. callURI = SIPURI.ParseSIPURIRelaxed(destination + "@" + m_sipServer); sipUsername = m_sipUsername; sipPassword = m_sipPassword; fromHeader = (new SIPFromHeader(m_sipFromName, new SIPURI(m_sipUsername, m_sipServer, null), null)).ToString(); } StatusMessage(this, $"Starting call to {callURI}."); var lookupResult = await Task.Run(() => { return(SIPDNSManager.ResolveSIPService(callURI, false)); }); if (lookupResult == null || lookupResult.LookupError != null) { StatusMessage(this, $"Call failed, could not resolve {callURI}."); } else { var dstEndpoint = lookupResult.GetSIPEndPoint(); StatusMessage(this, $"Call progressing, resolved {callURI} to {dstEndpoint}."); System.Diagnostics.Debug.WriteLine($"DNS lookup result for {callURI}: {dstEndpoint}."); SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, null, null); m_rtpMediaSessionManager.Create(dstEndpoint.Address.AddressFamily); m_rtpMediaSessionManager.RTPMediaSession.RemotePutOnHold += OnRemotePutOnHold; m_rtpMediaSessionManager.RTPMediaSession.RemoteTookOffHold += OnRemoteTookOffHold; await m_userAgent.InitiateCall(callDescriptor, m_rtpMediaSessionManager.RTPMediaSession); } }
public void UDPProtocolToStringTest() { Console.WriteLine("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name); SIPURI sipURI = new SIPURI(SIPSchemesEnum.sip, SIPEndPoint.ParseSIPEndPoint("127.0.0.1")); Console.WriteLine(sipURI.ToString()); Assert.IsTrue(sipURI.ToString() == "sip:127.0.0.1:5060", "The SIP URI was not ToString'ed correctly."); Console.WriteLine("-----------------------------------------"); }
private static ConcurrentQueue <RTPEvent> _dtmfEvents = new ConcurrentQueue <RTPEvent>(); // Add a DTMF event to this queue to have the it sent static void Main() { Console.WriteLine("SIPSorcery client user agent example."); Console.WriteLine("Press ctrl-c to exit."); // Plumbing code to facilitate a graceful exit. CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream. bool isCallHungup = false; bool hasCallFailed = false; AddConsoleLogger(); SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI); Log.LogInformation($"Call destination {callUri}."); // Set up a default SIP transport. var sipTransport = new SIPTransport(); int port = SIPConstants.DEFAULT_SIP_PORT + 1000; sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port))); // Uncomment this line to see each SIP message sent and received. EnableTraceLogs(sipTransport); // Send an OPTIONS request to determine the local IP address to use for the RTP socket. var optionsTask = SendOptionsTaskAsync(sipTransport, callUri); var result = Task.WhenAny(optionsTask, Task.Delay(SIP_REQUEST_TIMEOUT_MILLISECONDS)); result.Wait(); if (optionsTask.IsCompletedSuccessfully == false || optionsTask.Result == null) { Log.LogError($"OPTIONS request to {callUri} failed."); } else { IPAddress localIPAddress = optionsTask.Result; // Initialise an RTP session to receive the RTP packets from the remote SIP server. Socket rtpSocket = null; Socket controlSocket = null; NetServices.CreateRtpSocket(localIPAddress, 49000, 49100, false, out rtpSocket, out controlSocket); var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null); var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null); // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var uac = new SIPClientUserAgent(sipTransport); uac.CallTrying += (uac, resp) => { Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); }; uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); _remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body); Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}."); } else { Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); } }; // The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call. sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) => { if (sipRequest.Method == SIPMethodsEnum.BYE) { SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null); SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); byeTransaction.SendFinalResponse(byeResponse); if (uac.IsUACAnswered) { Log.LogInformation("Call was hungup by remote server."); isCallHungup = true; rtpCts.Cancel(); } } }; // It's a good idea to start the RTP receiving socket before the call request is sent. // A SIP server will generally start sending RTP as soon as it has processed the incoming call request and // being ready to receive will stop any ICMP error response being generated. Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts)); Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts)); // Start the thread that places the call. SIPCallDescriptor callDescriptor = new SIPCallDescriptor( SIPConstants.SIP_DEFAULT_USERNAME, null, callUri.ToString(), SIPConstants.SIP_DEFAULT_FROMURI, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpSocket.LocalEndPoint as IPEndPoint, RTPPayloadTypesEnum.PCMU).ToString(), null); uac.Call(callDescriptor); // Ctrl-c will gracefully exit the call at any point. Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e) { e.Cancel = true; rtpCts.Cancel(); }; // At this point the call has been initiated and everything will be handled in an event handler or on the RTP // receive task. The code below is to gracefully exit. Task.Delay(3000).Wait(); // Add some DTMF events to the queue. These will be transmitted by the SendRtp thread. _dtmfEvents.Enqueue(new RTPEvent(0x05, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID)); Task.Delay(2000, rtpCts.Token).Wait(); _dtmfEvents.Enqueue(new RTPEvent(0x09, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID)); Task.Delay(2000, rtpCts.Token).Wait(); _dtmfEvents.Enqueue(new RTPEvent(0x02, false, RTPEvent.DEFAULT_VOLUME, 1200, DTMF_EVENT_PAYLOAD_ID)); Task.Delay(2000, rtpCts.Token).Wait(); Log.LogInformation("Exiting..."); rtpCts.Cancel(); rtpSocket?.Close(); controlSocket?.Close(); if (!isCallHungup && uac != null) { if (uac.IsUACAnswered) { Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}."); uac.Hangup(); } else if (!hasCallFailed) { Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}."); uac.Cancel(); } // Give the BYE or CANCEL request time to be transmitted. Log.LogInformation("Waiting 1s for call to clean up..."); Task.Delay(1000).Wait(); } } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } }
public string ToXMLNoParent() { string registrarStr = (m_registerServer != null) ? m_registerServer.ToString() : null; string registerContactStr = (m_registerContact != null) ? SafeXML.MakeSafeXML(m_registerContact.ToString()) : null; string providerXML = " <id>" + m_id + "</id>" + m_newLine + " <providertype>" + m_providerType + "<providertype>" + m_newLine + " <owner>" + m_owner + "</owner>" + m_newLine + " <adminmemberid>" + AdminMemberId + "</adminmemberid>" + m_newLine + " <providername>" + m_providerName + "</providername>" + m_newLine + " <providerusername>" + m_providerUsername + "</providerusername>" + m_newLine + " <providerpassword>" + m_providerPassword + "</providerpassword>" + m_newLine + " <providerserver>" + SafeXML.MakeSafeXML(m_providerServer.ToString()) + "</providerserver>" + m_newLine + " <providerauthusername>" + m_providerAuthUsername + "</providerauthusername>" + m_newLine + " <provideroutboundproxy>" + SafeXML.MakeSafeXML(m_providerOutboundProxy) + "</provideroutboundproxy>" + m_newLine + " <providerfrom>" + SafeXML.MakeSafeXML(m_providerFrom) + "</providerfrom>" + m_newLine + " <customheaders>" + SafeXML.MakeSafeXML(m_customHeaders) + "</customheaders>" + m_newLine + " <registercontact>" + registerContactStr + "</registercontact>" + m_newLine + " <registerexpiry>" + m_registerExpiry + "</registerexpiry>" + m_newLine + " <registerserver>" + registrarStr + "</registerserver>" + m_newLine + " <registerrealm>" + SafeXML.MakeSafeXML(m_registerRealm) + "</registerrealm>" + m_newLine + " <registerenabled>" + m_registerEnabled + "</registerenabled>" + m_newLine + " <registerenabledadmin>" + m_registerAdminEnabled + "</registerenabledadmin>" + m_newLine + " <registerdisabledreason>" + SafeXML.MakeSafeXML(m_registerDisabledReason) + "</registerdisabledreason>" + m_newLine + " <inserted>" + m_inserted.ToString("o") + "</inserted>" + m_newLine + " <lastupdate>" + m_lastUpdate.ToString("o") + "</lastupdate>" + m_newLine + " <isreadonly>" + m_isReadOnly + "</isreadonly>" + m_newLine + " <sendmwisubscribe>" + m_sendMWISubscribe + "</sendmwisubscribe>" + m_newLine; return(providerXML); }
/// <summary> /// Puts the dialog participant information to an XML element. /// </summary> /// <param name="nodeName">A participant can represent a local or remote party, the node name needs to be set to either "local" or "remote".</param> /// <returns>An XML element representing the dialog participant.</returns> public XElement ToXML(string nodeName) { XNamespace ns = m_dialogXMLNS; XNamespace ss = m_sipsorceryXMLNS; XElement participantElement = new XElement(ns + nodeName); if (URI != null) { XElement identityElement = new XElement(ns + "identity", URI.ToString()); if (!DisplayName.IsNullOrBlank()) { identityElement.Add(new XAttribute("display-name", DisplayName)); } participantElement.Add(identityElement); } if (TargetURI != null) { XElement targetElement = new XElement(ns + "target", new XAttribute("uri", TargetURI.ToString())); participantElement.Add(targetElement); } if (CSeq > 0) { XElement cseqElement = new XElement(ns + "cseq", CSeq); participantElement.Add(cseqElement); } if (!SwitchboardLineName.IsNullOrBlank()) { XElement switchLineNameElement = new XElement(ss + "switchboardlinename", SwitchboardLineName); participantElement.Add(switchLineNameElement); } if (!CRMPersonName.IsNullOrBlank()) { XElement crmPersonNameElement = new XElement(ss + "crmpersonname", CRMPersonName); participantElement.Add(crmPersonNameElement); } if (!CRMCompanyName.IsNullOrBlank()) { XElement crmCompanyNameElement = new XElement(ss + "crmcompanyname", CRMCompanyName); participantElement.Add(crmCompanyNameElement); } if (!CRMPictureURL.IsNullOrBlank()) { XElement crmPictureURLElement = new XElement(ss + "crmpictureurl", CRMPictureURL); participantElement.Add(crmPictureURLElement); } return(participantElement); }
static async Task 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; 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); // Get the IP address the RTP will be sent from. While we can listen on IPAddress.Any | IPv6Any // we can't put 0.0.0.0 or [::0] in the SDP or the callee will treat our RTP stream as inactive. var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false); Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}."); var dstAddress = lookupResult.GetSIPEndPoint().Address; // Initialise an RTP session to receive the RTP packets from the remote SIP server. var audioOptions = new AudioOptions { AudioSource = AudioSourcesEnum.Microphone, AudioCodecs = new List <SDPMediaFormatsEnum> { SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.PCMU } }; var rtpSession = new RtpAVSession(audioOptions, null); var offerSDP = await rtpSession.createOffer(new RTCOfferOptions { RemoteSignallingAddress = dstAddress }); // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var uac = new SIPClientUserAgent(sipTransport); uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); rtpSession.setRemoteDescription(new RTCSessionDescription { type = RTCSdpType.answer, sdp = SDP.ParseSDPDescription(resp.Body) }); rtpSession.Start(); } else { Log.LogWarning($"{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); uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}"); // Ctrl-c will gracefully exit the call at any point. Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e) { e.Cancel = true; exitMre.Set(); }; // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed. exitMre.WaitOne(); Log.LogInformation("Exiting..."); rtpSession.CloseSession(null); if (!isCallHungup && uac != null) { if (uac.IsUACAnswered) { Log.LogInformation($"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(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } }
public void ReferOutOfDialog(SIPURI fromUri, SIPURI toUri, SIPURI referToUri, ReplacesCallDescriptor sipReplacesCallDescriptor) { try { m_sipCallDescriptor = new SIPCallDescriptor(null,toUri.ToString(),fromUri.ToString(),null,null); m_sipCallDescriptor.Gruu = toUri.Parameters.Get(SIPCallDescriptor.GRUU_KEY); m_sipCallDescriptor.ReplacesCall = sipReplacesCallDescriptor; SIPURI callURI = SIPURI.ParseSIPURI(m_sipCallDescriptor.Uri); // If the outbound proxy is a loopback address, as it will normally be for local deployments, then it cannot be overriden. if (m_outboundProxy != null && IPAddress.IsLoopback(m_outboundProxy.Address)) { m_serverEndPoint = m_outboundProxy; } else if (!m_sipCallDescriptor.ProxySendFrom.IsNullOrBlank()) { // If the binding has a specific proxy end point sent then the request needs to be forwarded to the proxy's default end point for it to take care of. SIPEndPoint outboundProxyEndPoint = SIPEndPoint.ParseSIPEndPoint(m_sipCallDescriptor.ProxySendFrom); m_outboundProxy = new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(outboundProxyEndPoint.Address, m_defaultSIPPort)); m_serverEndPoint = m_outboundProxy; Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "SIPReferClientUserAgent refer request using alternate outbound proxy of " + m_outboundProxy + ".", Owner)); } else if (m_outboundProxy != null) { // Using the system outbound proxy only, no additional user routing requirements. m_serverEndPoint = m_outboundProxy; } // A custom route set may have been specified for the call. if (m_sipCallDescriptor.RouteSet != null && m_sipCallDescriptor.RouteSet.IndexOf(OUTBOUNDPROXY_AS_ROUTESET_CHAR) != -1) { try { RouteSet = new SIPRouteSet(); RouteSet.PushRoute(new SIPRoute(m_sipCallDescriptor.RouteSet, true)); } catch { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Error an outbound proxy value was not recognised in SIPReferClientUserAgent refer request. " + m_sipCallDescriptor.RouteSet + ".", Owner)); } } // No outbound proxy, determine the forward destination based on the SIP request. if (m_serverEndPoint == null) { SIPDNSLookupResult lookupResult = null; if (RouteSet == null || RouteSet.Length == 0) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Attempting to resolve " + callURI.Host + ".", Owner)); lookupResult = m_sipTransport.GetURIEndPoint(callURI, false); } else { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Route set for refer request " + RouteSet.ToString() + ".", Owner)); lookupResult = m_sipTransport.GetURIEndPoint(RouteSet.TopRoute.URI, false); } if (lookupResult.LookupError != null) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "DNS error resolving " + callURI.Host + ", " + lookupResult.LookupError + ". Refer request cannot proceed.", Owner)); } else { m_serverEndPoint = lookupResult.GetSIPEndPoint(); } } if (m_serverEndPoint != null) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Switching to " + SIPURI.ParseSIPURI(m_sipCallDescriptor.Uri).CanonicalAddress + " via " + m_serverEndPoint + ".", Owner)); m_localSIPEndPoint = m_sipTransport.GetDefaultSIPEndPoint(m_serverEndPoint); if (m_localSIPEndPoint == null) { throw new ApplicationException("The refer request could not locate an appropriate SIP transport channel for protocol " + callURI.Protocol + "."); } SIPRequest referRequest = GetReferRequest(m_localSIPEndPoint,referToUri, sipReplacesCallDescriptor); // Now that we have a destination socket create a new UAC transaction for forwarded leg of the call. m_serverTransaction = m_sipTransport.CreateNonInviteTransaction(referRequest, m_serverEndPoint, m_localSIPEndPoint, m_outboundProxy); m_serverTransaction.NonInviteTransactionFinalResponseReceived += m_serverTransaction_NonInviteTransactionFinalResponseReceived; m_serverTransaction.NonInviteTransactionTimedOut += m_serverTransaction_NonInviteTransactionTimedOut; m_serverTransaction.TransactionTraceMessage += TransactionTraceMessage; m_serverTransaction.SendReliableRequest(); } else { if (RouteSet == null || RouteSet.Length == 0) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Forward leg failed, could not resolve URI host " + callURI.Host, Owner)); FireReferFailed(this, "unresolvable destination " + callURI.Host); } else { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Forward leg failed, could not resolve top Route host " + RouteSet.TopRoute.Host, Owner)); FireReferFailed(this, "unresolvable destination " + RouteSet.TopRoute.Host); } } } catch (ApplicationException appExcp) { FireReferFailed(this, appExcp.Message); } catch (Exception excp) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.ReferUserAgentClient, SIPMonitorEventTypesEnum.Refer, "Exception UserAgentClient Call. " + excp.Message, Owner)); FireReferFailed(this, excp.Message); } }
private SIPRequest GetReferRequest(SIPEndPoint localSIPEndPoint, SIPURI referTo, ReplacesCallDescriptor sipReplacesCallDescriptor) { SIPRequest referRequest = new SIPRequest(SIPMethodsEnum.REFER, m_sipCallDescriptor.Uri); SIPFromHeader referFromHeader = SIPFromHeader.ParseFromHeader(m_sipCallDescriptor.From); SIPToHeader referToHeader = SIPToHeader.ParseToHeader(m_sipCallDescriptor.Uri); int cseq = ++CSeq; CallId = CallProperties.CreateNewCallId(); SIPHeader referHeader = new SIPHeader(referFromHeader, referToHeader, cseq, CallId); referHeader.CSeqMethod = SIPMethodsEnum.REFER; referRequest.Header = referHeader; referRequest.Header.Routes = RouteSet; referRequest.Header.ProxySendFrom = m_sipCallDescriptor.ProxySendFrom; if (sipReplacesCallDescriptor != null) { referTo.Headers.Set("Replaces", sipReplacesCallDescriptor.GetUrlEncodedReplacesUriHeader()); referRequest.Header.ReferTo = referTo.ToString(); } else { referRequest.Header.ReferTo = referTo.ToString(); } // "Requires" Header referRequest.Header.Require = (String.IsNullOrWhiteSpace(referRequest.Header.Require)) ? "replaces" : referRequest.Header.Require + ",replaces"; SIPViaHeader viaHeader = new SIPViaHeader(localSIPEndPoint, CallProperties.CreateBranchId()); referRequest.Header.Vias.PushViaHeader(viaHeader); return referRequest; }
static void Main() { Console.WriteLine("SIPSorcery client user agent example."); Console.WriteLine("Press ctrl-c to exit."); // Plumbing code to facilitate a graceful exit. CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream. bool isCallHungup = false; bool hasCallFailed = false; AddConsoleLogger(); SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI); Log.LogInformation($"Call destination {callUri}."); // Set up a default SIP transport. var sipTransport = new SIPTransport(); sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0))); // Un/comment this line to see/hide each SIP message sent and received. EnableTraceLogs(sipTransport); // Note this relies on the callURI host being an IP address. If it's a hostname a DNS lookup is required. IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(callUri.ToSIPEndPoint().Address); // Initialise an RTP session to receive the RTP packets from the remote SIP server. var rtpSession = new RTPMediaSession((int)SDPMediaFormatsEnum.PCMU, localIPAddress.AddressFamily); var offerSDP = rtpSession.GetSDP(localIPAddress); // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var uac = new SIPClientUserAgent(sipTransport); uac.CallTrying += (uac, resp) => { Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); }; uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); rtpSession.DestinationEndPoint = SDP.GetSDPRTPEndPoint(resp.Body); Log.LogDebug($"Remote RTP socket {rtpSession.DestinationEndPoint}."); } else { Log.LogWarning($"{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; rtpCts.Cancel(); } } }; // Wire up the RTP receive session to the default speaker. var(audioOutEvent, audioOutProvider) = GetAudioOutputDevice(); rtpSession.OnReceivedSampleReady += (sample) => { for (int index = 0; index < sample.Length; index++) { short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]); byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) }; audioOutProvider.AddSamples(pcmSample, 0, 2); } }; // Send audio packets (in this case silence) to the callee. Task.Run(() => SendSilence(rtpSession, rtpCts)); // Start the thread that places the call. SIPCallDescriptor callDescriptor = new SIPCallDescriptor( SIPConstants.SIP_DEFAULT_USERNAME, null, callUri.ToString(), SIPConstants.SIP_DEFAULT_FROMURI, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, offerSDP.ToString(), null); uac.Call(callDescriptor); // Ctrl-c will gracefully exit the call at any point. Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e) { e.Cancel = true; rtpCts.Cancel(); }; // Give the call some time to answer. Task.Delay(3000).Wait(); // Send some DTMF key presses via RTP events. var dtmf5 = new RTPEvent(0x05, false, RTPEvent.DEFAULT_VOLUME, 1200, RTPMediaSession.DTMF_EVENT_PAYLOAD_ID); rtpSession.SendDtmfEvent(dtmf5, rtpCts.Token).Wait(); Task.Delay(2000, rtpCts.Token).Wait(); var dtmf9 = new RTPEvent(0x09, false, RTPEvent.DEFAULT_VOLUME, 1200, RTPMediaSession.DTMF_EVENT_PAYLOAD_ID); rtpSession.SendDtmfEvent(dtmf9, rtpCts.Token).Wait(); Task.Delay(2000, rtpCts.Token).Wait(); var dtmf2 = new RTPEvent(0x02, false, RTPEvent.DEFAULT_VOLUME, 1200, RTPMediaSession.DTMF_EVENT_PAYLOAD_ID); rtpSession.SendDtmfEvent(dtmf2, rtpCts.Token).Wait(); Task.Delay(2000, rtpCts.Token).ContinueWith((task) => { }).Wait(); // Don't care about the exception if the cancellation token is set. Log.LogInformation("Exiting..."); rtpCts.Cancel(); audioOutEvent?.Stop(); rtpSession.CloseSession(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(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } }
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(LogEventLevel.Verbose); 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(); //sipTransport.PreferIPv6NameResolution = true; sipTransport.EnableTraceLogs(); var audioSession = new WindowsAudioEndPoint(new AudioEncoder()); audioSession.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMA || x.Codec == AudioCodecsEnum.PCMU); //audioSession.RestrictFormats(x => x.Codec == AudioCodecsEnum.G722); 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) { if (resp.Body != null) { var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body)); if (result == SetDescriptionResultEnum.OK) { await rtpSession.Start(); Log.LogInformation($"Remote SDP set from in progress response. RTP session started."); } } } }; uac.CallFailed += (uac, err, resp) => { Log.LogWarning($"Call attempt to {uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += async(iuac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); if (resp.Body != null) { var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body)); if (result == SetDescriptionResultEnum.OK) { await rtpSession.Start(); } else { Log.LogWarning($"Failed to set remote description {result}."); uac.Hangup(); } } else if (!rtpSession.IsStarted) { Log.LogWarning($"Failed to set get remote description in session progress or final response."); uac.Hangup(); } } else { Log.LogWarning($"{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); // 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(); } }
/// <summary> /// Places an outgoing SIP call. /// </summary> /// <param name="destination">The SIP URI to place a call to. The destination can be a full SIP URI in which case the all will /// be placed anonymously directly to that URI. Alternatively it can be just the user portion of a URI in which case it will /// be sent to the configured SIP server.</param> public async Task Call(string destination) { // Determine if this is a direct anonymous call or whether it should be placed using the pre-configured SIP server account. SIPURI callURI = null; string sipUsername = null; string sipPassword = null; string fromHeader = null; if (destination.Contains("@") || m_sipServer == null) { // Anonymous call direct to SIP server specified in the URI. callURI = SIPURI.ParseSIPURIRelaxed(destination); fromHeader = (new SIPFromHeader(m_sipFromName, SIPURI.ParseSIPURI(SIPFromHeader.DEFAULT_FROM_URI), null)).ToString(); } else { // This call will use the pre-configured SIP account. callURI = SIPURI.ParseSIPURIRelaxed(destination + "@" + m_sipServer); sipUsername = m_sipUsername; sipPassword = m_sipPassword; fromHeader = (new SIPFromHeader(m_sipFromName, new SIPURI(m_sipFromName, m_sipServer, null), null)).ToString(); } StatusMessage(this, $"Starting call to {callURI}."); var dstEndpoint = await SIPDns.ResolveAsync(callURI, false, _cts.Token); if (dstEndpoint == null) { StatusMessage(this, $"Call failed, could not resolve {callURI}."); } else { StatusMessage(this, $"Call progressing, resolved {callURI} to {dstEndpoint}."); System.Diagnostics.Debug.WriteLine($"DNS lookup result for {callURI}: {dstEndpoint}."); SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, null, null); callDescriptor.CallId = m_sipFromName; MediaSession = CreateMediaSession(); m_userAgent.RemotePutOnHold += OnRemotePutOnHold; m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold; Directory.CreateDirectory(SIPSoftPhoneState.OutputRecordingFolder); string guid = Guid.NewGuid().ToString(); _waveFile = new WaveFileWriter(Path.Combine(SIPSoftPhoneState.OutputRecordingFolder, $"{guid}.mp3"), _waveFormat); await m_userAgent.InitiateCallAsync(callDescriptor, MediaSession); } }
/// <summary> /// Places an outgoing SIP call. /// </summary> /// <param name="destination">The SIP URI to place a call to. The destination can be a full SIP URI in which case the all will /// be placed anonymously directly to that URI. Alternatively it can be just the user portion of a URI in which case it will /// be sent to the configured SIP server.</param> public async Task Call(string destination) { // Determine if this is a direct anonymous call or whether it should be placed using the pre-configured SIP server account. SIPURI callURI = null; string sipUsername = null; string sipPassword = null; string fromHeader = null; if (destination.Contains("@") || m_sipServer == null) { // Anonymous call direct to SIP server specified in the URI. callURI = SIPURI.ParseSIPURIRelaxed(destination); fromHeader = (new SIPFromHeader(m_sipFromName, SIPURI.ParseSIPURI(SIPFromHeader.DEFAULT_FROM_URI), null)).ToString(); } else { // This call will use the pre-configured SIP account. callURI = SIPURI.ParseSIPURIRelaxed(destination + "@" + m_sipServer); sipUsername = m_sipUsername; sipPassword = m_sipPassword; fromHeader = (new SIPFromHeader(m_sipFromName, new SIPURI(m_sipUsername, m_sipServer, null), null)).ToString(); } StatusMessage(this, $"Starting call to {callURI}."); var lookupResult = await Task.Run(() => { return(SIPDNSManager.ResolveSIPService(callURI, false)); }); if (lookupResult == null || lookupResult.LookupError != null) { StatusMessage(this, $"Call failed, could not resolve {callURI}."); } else { var dstEndpoint = lookupResult.GetSIPEndPoint(); StatusMessage(this, $"Call progressing, resolved {callURI} to {dstEndpoint}."); System.Diagnostics.Debug.WriteLine($"DNS lookup result for {callURI}: {dstEndpoint}."); SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, null, null); var audioSrcOpts = new AudioOptions { AudioSource = AudioSourcesEnum.CaptureDevice, OutputDeviceIndex = m_audioOutDeviceIndex }; var videoSrcOpts = new VideoOptions { VideoSource = VideoSourcesEnum.TestPattern, SourceFile = RtpAVSession.VIDEO_TESTPATTERN, SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND }; MediaSession = new RtpAVSession(audioSrcOpts, videoSrcOpts); m_userAgent.RemotePutOnHold += OnRemotePutOnHold; m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold; await m_userAgent.InitiateCallAsync(callDescriptor, MediaSession); } }
// TODO need to consolidate ResolveSIPService and ResolveAsync. public static SIPDNSLookupResult ResolveSIPService(SIPURI sipURI, bool async) { try { if (sipURI == null) { throw new ArgumentNullException("sipURI", "Cannot resolve SIP service on a null URI."); } if (IPSocket.TryParseIPEndPoint(sipURI.Host, out var ipEndPoint)) { // Target is an IP address, no DNS lookup required. SIPDNSLookupEndPoint sipLookupEndPoint = new SIPDNSLookupEndPoint(new SIPEndPoint(sipURI.Protocol, ipEndPoint), 0); SIPDNSLookupResult result = new SIPDNSLookupResult(sipURI); result.AddLookupResult(sipLookupEndPoint); return(result); } else { string host = sipURI.Host; int port = IPSocket.ParsePortFromSocket(sipURI.Host); bool explicitPort = (port != 0); if (!explicitPort) { port = (sipURI.Scheme == SIPSchemesEnum.sip) ? m_defaultSIPPort : m_defaultSIPSPort; } if (host.Contains(".") == false) { string hostOnly = IPSocket.ParseHostFromSocket(host); // If host is not fully qualified then assume there's no point using NAPTR or SRV record look ups and go straight to A's. if (hostOnly.ToLower() == System.Net.Dns.GetHostName()?.ToLower()) { // The lookup is for the current machine. var addressList = System.Net.Dns.GetHostEntry(System.Net.Dns.GetHostName()).AddressList; if (addressList?.Length == 0) { return(new SIPDNSLookupResult(sipURI, $"Failed to resolve local machine hostname.")); } else { // Preference for IPv4 IP address for local host anem lookup. IPAddress firstAddress = addressList.Where(x => x.AddressFamily == AddressFamily.InterNetwork).FirstOrDefault() ?? addressList.FirstOrDefault(); SIPEndPoint resultEp = new SIPEndPoint(sipURI.Protocol, new IPEndPoint(firstAddress, port)); return(new SIPDNSLookupResult(sipURI, resultEp)); } } else { return(DNSARecordLookup(hostOnly, port, async, sipURI)); } } else if (explicitPort) { // If target is a hostname with an explicit port then SIP lookup rules state to use DNS lookup for A or AAAA record. host = host.Substring(0, host.LastIndexOf(':')); return(DNSARecordLookup(host, port, async, sipURI)); } else { // Target is a hostname with no explicit port, use the whole NAPTR->SRV->A lookup procedure. SIPDNSLookupResult sipLookupResult = new SIPDNSLookupResult(sipURI); // Do without the NAPTR lookup for the time being. Very few organisations appear to use them and it can cost up to 2.5s to get a failed resolution. /*SIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Unknown, SIPMonitorEventTypesEnum.DNS, "SIP DNS full lookup requested for " + sipURI.ToString() + ".", null)); * DNSNAPTRRecordLookup(host, async, ref sipLookupResult); * if (sipLookupResult.Pending) * { * if (!m_inProgressSIPServiceLookups.Contains(sipURI.ToString())) * { * m_inProgressSIPServiceLookups.Add(sipURI.ToString()); * ThreadPool.QueueUserWorkItem(delegate { ResolveSIPService(sipURI, false); }); * } * return sipLookupResult; * }*/ DNSSRVRecordLookup(sipURI.Scheme, sipURI.Protocol, host, async, ref sipLookupResult); if (sipLookupResult.Pending) { //logger.LogDebug("SIPDNSManager SRV lookup for " + host + " is pending."); return(sipLookupResult); } else { //logger.LogDebug("SIPDNSManager SRV lookup for " + host + " is final."); // Add some custom logic to cope with sips SRV records using _sips._tcp (e.g. free.call.ciscospark.com). // By default only _sips._tls SRV records are checked for. THis block adds an additional check for _sips._tcp SRV records. //if ((sipLookupResult.SIPSRVResults == null || sipLookupResult.SIPSRVResults.Count == 0) && sipURI.Scheme == SIPSchemesEnum.sips) //{ // DNSSRVRecordLookup(sipURI.Scheme, SIPProtocolsEnum.tcp, host, async, ref sipLookupResult); // SIPDNSServiceResult nextSRVRecord = sipLookupResult.GetNextUnusedSRV(); // int lookupPort = (nextSRVRecord != null) ? nextSRVRecord.Port : port; // return DNSARecordLookup(nextSRVRecord, host, lookupPort, async, sipLookupResult.URI); //} //else //{ SIPDNSServiceResult nextSRVRecord = sipLookupResult.GetNextUnusedSRV(); int lookupPort = (nextSRVRecord != null) ? nextSRVRecord.Port : port; return(DNSARecordLookup(nextSRVRecord, host, lookupPort, async, sipLookupResult.URI)); //} } } } } catch (Exception excp) { logger.LogError("Exception SIPDNSManager ResolveSIPService (" + sipURI.ToString() + "). " + excp.Message); m_inProgressSIPServiceLookups.Remove(sipURI.ToString()); return(new SIPDNSLookupResult(sipURI, excp.Message)); } }
public void ProcessNotifyRequest(SIPRequest sipRequest) { try { // Hack to work around MWI request from callcentric not having a trailing CRLF and breaking some softphones like the Bria. if (sipRequest.Header.Event == MWI_EVENT_TYPE && sipRequest.Body.NotNullOrBlank() && sipRequest.Body.Substring(sipRequest.Body.Length - 2) != m_crlf) { sipRequest.Body += m_crlf; } string fromURI = (sipRequest.Header.From != null && sipRequest.Header.From.FromURI != null) ? sipRequest.Header.From.FromURI.ToString() : "unknown"; string domain = GetCanonicalDomain_External(sipRequest.URI.Host, true); if (domain != null) { SIPAccount sipAccount = GetSIPAccount_External(s => s.SIPUsername == sipRequest.URI.User && s.SIPDomain == domain); if (sipAccount != null) { List <SIPRegistrarBinding> bindings = GetSIPAccountBindings_External(b => b.SIPAccountId == sipAccount.Id, null, 0, MAX_FORWARD_BINDINGS); if (bindings != null) { foreach (SIPRegistrarBinding binding in bindings) { SIPURI dstURI = binding.MangledContactSIPURI; SIPEndPoint localSIPEndPoint = (m_outboundProxy != null) ? m_sipTransport.GetDefaultSIPEndPoint(m_outboundProxy.Protocol) : m_sipTransport.GetDefaultSIPEndPoint(dstURI.Protocol); SIPEndPoint dstSIPEndPoint = null; // If the outbound proxy is a loopback address, as it will normally be for local deployments, then it cannot be overriden. if (m_outboundProxy != null && IPAddress.IsLoopback(m_outboundProxy.Address)) { dstSIPEndPoint = m_outboundProxy; } else if (binding.ProxySIPEndPoint != null) { // If the binding has a specific proxy end point sent then the request needs to be forwarded to the proxy's default end point for it to take care of. dstSIPEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(binding.ProxySIPEndPoint.Address, m_defaultSIPPort)); } else if (m_outboundProxy != null) { dstSIPEndPoint = m_outboundProxy; } else { SIPDNSLookupResult lookupResult = m_sipTransport.GetURIEndPoint(dstURI, false); if (lookupResult.LookupError != null) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "A NOTIFY request from " + fromURI + " was not forwarded due to DNS failure for " + dstURI.Host + ", " + lookupResult.LookupError + ".", sipAccount.Owner)); } else { dstSIPEndPoint = lookupResult.GetSIPEndPoint(); } } if (dstSIPEndPoint != null) { // Rather than create a brand new request copy the received one and modify the headers that need to be unique. SIPRequest notifyRequest = sipRequest.Copy(); notifyRequest.URI = dstURI; notifyRequest.Header.Contact = SIPContactHeader.CreateSIPContactList(new SIPURI(dstURI.Scheme, localSIPEndPoint)); notifyRequest.Header.To = new SIPToHeader(null, dstURI, null); notifyRequest.Header.CallId = CallProperties.CreateNewCallId(); SIPViaHeader viaHeader = new SIPViaHeader(localSIPEndPoint, CallProperties.CreateBranchId()); notifyRequest.Header.Vias = new SIPViaSet(); notifyRequest.Header.Vias.PushViaHeader(viaHeader); // If the binding has a proxy socket defined set the header to ask the upstream proxy to use it. if (binding.ProxySIPEndPoint != null) { notifyRequest.Header.ProxySendFrom = binding.ProxySIPEndPoint.ToString(); // If the binding has a specific proxy end point sent then the request needs to be forwarded to the proxy's default end point for it to take care of. dstSIPEndPoint = new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(binding.ProxySIPEndPoint.Address, m_defaultSIPPort)); } Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "Forwarding NOTIFY request from " + fromURI + " to registered binding at " + dstURI.ToString() + ", proxy " + dstSIPEndPoint.ToString() + ".", sipAccount.Owner)); SIPNonInviteTransaction notifyTransaction = m_sipTransport.CreateNonInviteTransaction(notifyRequest, dstSIPEndPoint, localSIPEndPoint, dstSIPEndPoint); notifyTransaction.SendReliableRequest(); } else { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "A NOTIFY request from " + fromURI + " was not forwarded as no destination end point was resolved.", sipAccount.Owner)); } } // Send OK response to server. SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); m_sipTransport.SendResponse(okResponse); } else { // Send unavailable response to server. Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "NOTIFY request from " + fromURI + " for " + sipAccount.SIPUsername + "@" + sipAccount.SIPDomain + " but no bindings available, responding with temporarily unavailable.", sipAccount.Owner)); SIPResponse notAvailableResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.TemporarilyUnavailable, null); m_sipTransport.SendResponse(notAvailableResponse); } } else { // Send Not found response to server. Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "NOTIFY request from " + fromURI + " for " + sipRequest.URI.ToString() + " but no matching SIP account, responding with not found.", null)); SIPResponse notFoundResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, null); m_sipTransport.SendResponse(notFoundResponse); } } else { // Send Not Serviced response to server. Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.MWI, "NOTIFY request from " + fromURI + " for a non-serviced domain responding with not found.", null)); SIPResponse notServicedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain not serviced"); m_sipTransport.SendResponse(notServicedResponse); } } catch (Exception excp) { logger.Error("Exception SIPNotifyManager ProcessNotifyRequest. " + excp.Message); } }
static void Main(string[] args) { Console.WriteLine("SIPSorcery client user agent example."); Console.WriteLine("Press ctrl-c to exit."); // Plumbing code to facilitate a graceful exit. CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream. bool isCallHungup = false; bool hasCallFailed = false; // Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required. var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory(); var loggerConfig = new LoggerConfiguration() .Enrich.FromLogContext() .MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug) .WriteTo.Console() .CreateLogger(); loggerFactory.AddSerilog(loggerConfig); SIPSorcery.Sys.Log.LoggerFactory = loggerFactory; SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI); if (args != null && args.Length > 0) { if (!SIPURI.TryParse(args[0])) { Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}"); } else { callUri = SIPURI.ParseSIPURIRelaxed(args[0]); } } Log.LogInformation($"Call destination {callUri}."); // Set up a default SIP transport. var sipTransport = new SIPTransport(); int port = SIPConstants.DEFAULT_SIP_PORT + 1000; IPAddress localAddress = sipTransport.GetLocalAddress(IPAddress.Parse("8.8.8.8")); sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(localAddress, port))); //sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port))); //sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, port))); //EnableTraceLogs(sipTransport); // Select the IP address to use for RTP based on the destination SIP URI. var endPointForCall = callUri.ToSIPEndPoint() == null ? sipTransport.GetDefaultSIPEndPoint(callUri.Protocol) : sipTransport.GetDefaultSIPEndPoint(callUri.ToSIPEndPoint()); // Initialise an RTP session to receive the RTP packets from the remote SIP server. Socket rtpSocket = null; Socket controlSocket = null; // TODO (find something better): If the SIP endpoint is using 0.0.0.0 for SIP use loopback for RTP. IPAddress rtpAddress = localAddress; NetServices.CreateRtpSocket(rtpAddress, 49000, 49100, false, out rtpSocket, out controlSocket); var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null); var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null); // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var uac = new SIPClientUserAgent(sipTransport); uac.CallTrying += (uac, resp) => { Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); }; uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); _remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body); Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}."); } else { Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); } }; // The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call. sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) => { if (sipRequest.Method == SIPMethodsEnum.BYE) { SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null); SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); byeTransaction.SendFinalResponse(byeResponse); if (uac.IsUACAnswered) { Log.LogInformation("Call was hungup by remote server."); isCallHungup = true; rtpCts.Cancel(); } } }; // It's a good idea to start the RTP receiving socket before the call request is sent. // A SIP server will generally start sending RTP as soon as it has processed the incoming call request and // being ready to receive will stop any ICMP error response being generated. Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts)); Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts)); // Start the thread that places the call. SIPCallDescriptor callDescriptor = new SIPCallDescriptor( SIPConstants.SIP_DEFAULT_USERNAME, null, callUri.ToString(), SIPConstants.SIP_DEFAULT_FROMURI, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(), null); uac.Call(callDescriptor); // Ctrl-c will gracefully exit the call at any point. Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e) { e.Cancel = true; rtpCts.Cancel(); }; // At this point the call has been initiated and everything will be handled in an event handler or on the RTP // receive task. The code below is to gracefully exit. // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed. rtpCts.Token.WaitHandle.WaitOne(); Log.LogInformation("Exiting..."); rtpSocket?.Close(); controlSocket?.Close(); if (!isCallHungup && uac != null) { if (uac.IsUACAnswered) { Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}."); uac.Hangup(); } else if (!hasCallFailed) { Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}."); uac.Cancel(); } // Give the BYE or CANCEL request time to be transmitted. Log.LogInformation("Waiting 1s for call to clean up..."); Task.Delay(1000).Wait(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } }
// TODO: Remove this nasty code duplication with ResolveSIPService! public static async Task <SIPDNSLookupResult> ResolveAsync(SIPURI sipURI, bool?preferIPv6 = null) { try { if (sipURI == null) { throw new ArgumentNullException("sipURI", "Cannot resolve SIP service on a null URI."); } if (IPSocket.TryParseIPEndPoint(sipURI.MAddrOrHost, out var ipEndPoint)) { // Target is an IP address, no DNS lookup required. SIPDNSLookupEndPoint sipLookupEndPoint = new SIPDNSLookupEndPoint(new SIPEndPoint(sipURI.Protocol, ipEndPoint), 0); SIPDNSLookupResult result = new SIPDNSLookupResult(sipURI); result.AddLookupResult(sipLookupEndPoint); return(result); } else { string host = sipURI.MAddrOrHostAddress; //int port = IPSocket.ParsePortFromSocket(host); int port = sipURI.Protocol != SIPProtocolsEnum.tls ? m_defaultSIPPort : m_defaultSIPSPort; if (preferIPv6 == null) { preferIPv6 = SIPDNSManager.PreferIPv6NameResolution; } bool explicitPort = false; try { //Parse returns true if sipURI.Host can be parsed as an ipaddress bool parseresult = SIPSorcery.Sys.IPSocket.Parse(sipURI.MAddrOrHost, out host, out port); explicitPort = port >= 0; if (!explicitPort) { //port = (sipURI.Scheme == SIPSchemesEnum.sip) ? m_defaultSIPPort : m_defaultSIPSPort; port = sipURI.Protocol != SIPProtocolsEnum.tls ? m_defaultSIPPort : m_defaultSIPSPort; } if (parseresult == true) { IPAddress hostIP = IPAddress.Parse(host); SIPDNSLookupEndPoint sipLookupEndPoint = new SIPDNSLookupEndPoint(new SIPEndPoint(sipURI.Protocol, new IPEndPoint(hostIP, port)), 0); SIPDNSLookupResult result = new SIPDNSLookupResult(sipURI); result.AddLookupResult(sipLookupEndPoint); return(result); } } catch (Exception ex) { SIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Unknown, SIPMonitorEventTypesEnum.DNS, "Exception SIPDNSManager ResolveSIPService (" + sipURI.ToString() + "). " + ex, null)); //rj2: if there is a parsing exception, then fallback to original sipsorcery parsing mechanism port = IPSocket.ParsePortFromSocket(sipURI.Host); explicitPort = port > 0; //if(Regex.Match(host, @"^\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}$").Success) if (SIPSorcery.Sys.IPSocket.IsIPAddress(host)) { // Target is an IP address, no DNS lookup required. IPAddress hostIP = IPAddress.Parse(host); SIPDNSLookupEndPoint sipLookupEndPoint = new SIPDNSLookupEndPoint(new SIPEndPoint(sipURI.Protocol, new IPEndPoint(hostIP, port)), 0); SIPDNSLookupResult result = new SIPDNSLookupResult(sipURI); result.AddLookupResult(sipLookupEndPoint); return(result); } } if (!explicitPort) { //port = (sipURI.Scheme == SIPSchemesEnum.sip) ? m_defaultSIPPort : m_defaultSIPSPort; port = sipURI.Protocol != SIPProtocolsEnum.tls ? m_defaultSIPPort : m_defaultSIPSPort; } if (host.Contains(".") == false) { string hostOnly = IPSocket.ParseHostFromSocket(host); // If host is not fully qualified then assume there's no point using NAPTR or SRV record look ups and go straight to A's. if (hostOnly.ToLower() == System.Net.Dns.GetHostName()?.ToLower()) { // The lookup is for the current machine. var addressList = System.Net.Dns.GetHostEntry(System.Net.Dns.GetHostName()).AddressList; if (addressList?.Length == 0) { return(new SIPDNSLookupResult(sipURI, $"Failed to resolve local machine hostname.")); } else { // Preference for IPv4 IP address for local host name lookup. IPAddress firstAddress = addressList.Where(x => x.AddressFamily == (preferIPv6 == true ? AddressFamily.InterNetworkV6 : AddressFamily.InterNetwork)).FirstOrDefault() ?? addressList.FirstOrDefault(); SIPEndPoint resultEp = new SIPEndPoint(sipURI.Protocol, new IPEndPoint(firstAddress, port)); return(new SIPDNSLookupResult(sipURI, resultEp)); } } else { return(await Task.Run(() => DNSNameRecordLookup(hostOnly, port, false, sipURI, null, preferIPv6))); } } else if (explicitPort) { // If target is a hostname with an explicit port then SIP lookup rules state to use DNS lookup for A or AAAA record. host = host.Substring(0, host.LastIndexOf(':')); return(await Task.Run(() => DNSNameRecordLookup(host, port, false, sipURI, null, preferIPv6))); } else { // Target is a hostname with no explicit port, use the whole NAPTR->SRV->A lookup procedure. SIPDNSLookupResult sipLookupResult = new SIPDNSLookupResult(sipURI); // Do without the NAPTR lookup for the time being. Very few organisations appear to use them and it can cost up to 2.5s to get a failed resolution. //rj2: uncomment this section for telekom 1TR118 lookup SIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Unknown, SIPMonitorEventTypesEnum.DNS, "SIP DNS full lookup requested for " + sipURI.ToString() + ".", null)); DNSNAPTRRecordLookup(host, false, ref sipLookupResult); if (sipLookupResult.Pending) { if (!m_inProgressSIPServiceLookups.Contains(sipURI.ToString())) { m_inProgressSIPServiceLookups.Add(sipURI.ToString()); //ThreadPool.QueueUserWorkItem(delegate { ResolveSIPService(sipURI, false); }); System.Threading.ThreadPool.QueueUserWorkItem((obj) => ResolveSIPService(sipURI, false, preferIPv6)); } return(sipLookupResult); } return(await Task.Run(() => { DNSSRVRecordLookup(sipURI.Scheme, sipURI.Protocol, host, false, ref sipLookupResult); if (sipLookupResult.Pending) { //logger.LogDebug("SIPDNSManager SRV lookup for " + host + " is pending."); return sipLookupResult; } else { //logger.LogDebug("SIPDNSManager SRV lookup for " + host + " is final."); // Add some custom logic to cope with sips SRV records using _sips._tcp (e.g. free.call.ciscospark.com). // By default only _sips._tls SRV records are checked for. THis block adds an additional check for _sips._tcp SRV records. //if ((sipLookupResult.SIPSRVResults == null || sipLookupResult.SIPSRVResults.Count == 0) && sipURI.Scheme == SIPSchemesEnum.sips) //{ // DNSSRVRecordLookup(sipURI.Scheme, SIPProtocolsEnum.tcp, host, async, ref sipLookupResult); // SIPDNSServiceResult nextSRVRecord = sipLookupResult.GetNextUnusedSRV(); // int lookupPort = (nextSRVRecord != null) ? nextSRVRecord.Port : port; // return DNSARecordLookup(nextSRVRecord, host, lookupPort, async, sipLookupResult.URI); //} //else //{ // SIPDNSServiceResult nextSRVRecord = sipLookupResult.GetNextUnusedSRV(); // int lookupPort = (nextSRVRecord != null) ? nextSRVRecord.Port : port; // return DNSARecordLookup(nextSRVRecord, host, lookupPort, false, sipLookupResult.URI); //} // The SRV record can indicate the SIP server is using a non-standard port. SIPDNSServiceResult nextSRVRecord = sipLookupResult.GetNextUnusedSRV(); int lookupPort = (nextSRVRecord != null) ? nextSRVRecord.Port : port; return DNSNameRecordLookup(host, lookupPort, false, sipLookupResult.URI, ref sipLookupResult, preferIPv6); } })); } } } catch (Exception excp) { logger.LogError("Exception SIPDNSManager ResolveSIPService (" + sipURI.ToString() + "). " + excp.Message); m_inProgressSIPServiceLookups.Remove(sipURI.ToString()); return(new SIPDNSLookupResult(sipURI, excp.Message)); } }
private void ProcessRedirect(ISIPClientUserAgent answeredUAC, SIPResponse answeredResponse) { try { SIPURI redirectURI = answeredResponse.Header.Contact[0].ContactURI; if (redirectURI == null) { FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Redirect target could not be determined from redirect response, ignoring.", m_username)); } else { //string canincalDomain = m_dialPlanContext.GetCanonicalDomain_External(redirectURI.Host, false); if (answeredUAC.CallDescriptor.RedirectMode == SIPCallRedirectModesEnum.NewDialPlan) { m_dialPlanContext.ExecuteDialPlanForRedirect(answeredResponse); } else if (answeredUAC.CallDescriptor.RedirectMode == SIPCallRedirectModesEnum.Manual) { FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Redirect response with URI " + redirectURI.ToString() + " was not acted on as redirect mode set to manual in dial string.", m_username)); CallLegCompleted(); } else { // The redirect was not explicitly allowed so will be processed as an anonymous call. FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Redirect response to " + redirectURI.ToString() + " accepted.", m_username)); var redirectCalls = m_dialStringParser.GetForwardsForRedirect(redirectURI, answeredUAC.CallDescriptor); //SIPCallDescriptor redirectCallDescriptor = answeredUAC.CallDescriptor.CopyOf(); //redirectCallDescriptor.Uri = redirectURI.ToString(); //StartNewCallAsync(redirectCallDescriptor); if (redirectCalls != null && redirectCalls.Count > 0) { foreach (var redirectCall in redirectCalls) { StartNewCallAsync(redirectCall); } } } //else if (answeredUAC.CallDescriptor.RedirectMode == SIPCallRedirectModesEnum.Replace) //{ // // In the Replace redirect mode the existing dialplan execution needs to be cancelled and the single redirect call be used to replace it. // FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Redirect response rejected as Replace mode not yet implemented.", m_username)); // CallLegCompleted(); //} //else //{ // FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Redirect response with URI " + redirectURI.ToString() + " was not acted on as not enabled in dial string.", m_username)); // CallLegCompleted(); //} // A redirect response was received. Create a new call leg(s) using the SIP URIs in the contact header of the response. //if (m_dialStringParser != null) //{ // // If there is a dial string parser available it will be used to generate a list of call destination from the redirect URI. // SIPCallDescriptor redirectCallDescriptor = answeredUAC.CallDescriptor.CopyOf(); // Queue<List<SIPCallDescriptor>> redirectQueue = m_dialStringParser.ParseDialString(DialPlanContextsEnum.Script, null, redirectURI.ToString(), redirectCallDescriptor.CustomHeaders, // redirectCallDescriptor.ContentType, redirectCallDescriptor.Content, null, redirectCallDescriptor.FromDisplayName, redirectCallDescriptor.FromURIUsername, redirectCallDescriptor.FromURIHost, null, CustomerServiceLevels.None); // if (redirectQueue != null && redirectQueue.Count > 0) // { // // Only the first list in the queue is used (and there should only be a single list since it's generated from a redirect SIP URI and not // // a full dial string). // List<SIPCallDescriptor> callDescriptors = redirectQueue.Dequeue(); // for (int index = 0; index < callDescriptors.Count; index++) // { // callDescriptors[index].MangleIPAddress = redirectCallDescriptor.MangleIPAddress; // callDescriptors[index].MangleResponseSDP = redirectCallDescriptor.MangleResponseSDP; // callDescriptors[index].TransferMode = redirectCallDescriptor.TransferMode; // } // Start(callDescriptors); // } // else // { // FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "A redirect response to " + redirectURI.ToString() + " did not generate any new call leg destinations.", m_username)); // } //} //else //{ // FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Redirect response to " + redirectURI.ToString() + " accepted.", m_username)); // SIPCallDescriptor redirectCallDescriptor = answeredUAC.CallDescriptor.CopyOf(); // redirectCallDescriptor.Uri = redirectURI.ToString(); // StartNewCallAsync(redirectCallDescriptor); //} } } catch (Exception excp) { logger.Error("Exception ForkCall ProcessRedirect. " + excp.Message); } }
private Task <SocketError> ByeServerFinalResponseReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPTransaction sipTransaction, SIPResponse sipResponse) { try { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.UserAgentServer, SIPMonitorEventTypesEnum.DialPlan, "Response " + sipResponse.StatusCode + " " + sipResponse.ReasonPhrase + " for " + sipTransaction.TransactionRequest.URI.ToString() + ".", Owner)); SIPNonInviteTransaction byeTransaction = sipTransaction as SIPNonInviteTransaction; if ((sipResponse.Status == SIPResponseStatusCodesEnum.ProxyAuthenticationRequired || sipResponse.Status == SIPResponseStatusCodesEnum.Unauthorised) && SIPAccount != null) { // Resend BYE with credentials. SIPAuthorisationDigest authRequest = sipResponse.Header.AuthenticationHeader.SIPDigest; SIPURI contactUri = sipResponse.Header.Contact.Any() ? sipResponse.Header.Contact[0].ContactURI : sipResponse.Header.From.FromURI; authRequest.SetCredentials(SIPAccount.SIPUsername, SIPAccount.SIPPassword, contactUri.ToString(), SIPMethodsEnum.BYE.ToString()); SIPRequest authByeRequest = byeTransaction.TransactionRequest; authByeRequest.Header.AuthenticationHeader = new SIPAuthenticationHeader(authRequest); authByeRequest.Header.AuthenticationHeader.SIPDigest.Response = authRequest.Digest; authByeRequest.Header.Vias.TopViaHeader.Branch = CallProperties.CreateBranchId(); authByeRequest.Header.CSeq = authByeRequest.Header.CSeq + 1; SIPNonInviteTransaction authByeTransaction = new SIPNonInviteTransaction(m_sipTransport, authByeRequest, null); authByeTransaction.SendRequest(); } return(Task.FromResult(SocketError.Success)); } catch (Exception excp) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.UserAgentClient, SIPMonitorEventTypesEnum.Error, "Exception ByServerFinalResponseReceived. " + excp.Message, Owner)); return(Task.FromResult(SocketError.Fault)); } }
/// <summary> /// Updates the bindings list for a registrar's address-of-records. /// </summary> /// <param name="proxyEndPoint">If the request arrived at this registrar via a proxy then this will contain the end point of the proxy.</param> /// <param name="uacRecvdEndPoint">The public end point the UAC REGISTER request was deemded to have originated from.</param> /// <param name="registrarEndPoint">The registrar end point the registration request was received on.</param> /// <param name="maxAllowedExpiry">The maximum allowed expiry that can be granted to this binding request.</param> /// <returns>If the binding update was successful the expiry time for it is returned otherwise 0.</returns> public List <SIPRegistrarBinding> UpdateBindings( SIPAccount sipAccount, SIPEndPoint proxySIPEndPoint, SIPEndPoint remoteSIPEndPoint, SIPEndPoint registrarSIPEndPoint, List <SIPContactHeader> contactHeaders, string callId, int cseq, //int contactHeaderExpiresValue, int expiresHeaderValue, string userAgent, out SIPResponseStatusCodesEnum responseStatus, out string responseMessage) { //logger.Debug("UpdateBinding " + bindingURI.ToString() + "."); int maxAllowedExpiry = (m_userAgentConfigs != null) ? m_userAgentConfigs.GetMaxAllowedExpiry(userAgent) : SIPUserAgentConfiguration.DEFAULT_MAX_EXPIRY_SECONDS; responseMessage = null; string sipAccountAOR = sipAccount.SIPUsername + "@" + sipAccount.SIPDomain; responseStatus = SIPResponseStatusCodesEnum.Ok; try { userAgent = (userAgent != null && userAgent.Length > MAX_USERAGENT_LENGTH) ? userAgent.Substring(0, MAX_USERAGENT_LENGTH) : userAgent; List <SIPRegistrarBinding> bindings = m_bindingsPersistor.Get(b => b.SIPAccountId == sipAccount.Id, null, 0, Int32.MaxValue); foreach (SIPContactHeader contactHeader in contactHeaders) { SIPURI bindingURI = contactHeader.ContactURI.CopyOf(); int contactHeaderExpiresValue = contactHeader.Expires; int bindingExpiry = 0; if (bindingURI.Host == m_sipRegisterRemoveAll) { if (contactHeaders.Count > 1) { // If a register request specifies remove all it cannot contain any other binding requests. FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingRemoval, "Remove all bindings requested for " + sipAccountAOR + " but mutliple bindings specified, rejecting as a bad request.", sipAccount.SIPUsername)); responseStatus = SIPResponseStatusCodesEnum.BadRequest; break; } #region Process remove all bindings. if (expiresHeaderValue == 0) { // Removing all bindings for user. FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingRemoval, "Remove all bindings requested for " + sipAccountAOR + ".", sipAccount.SIPUsername)); // Mark all the current bindings as expired. if (bindings != null && bindings.Count > 0) { for (int index = 0; index < bindings.Count; index++) { bindings[index].RemovalReason = SIPBindingRemovalReason.ClientExpiredAll; bindings[index].Expiry = 0; m_bindingsPersistor.Update(bindings[index]); // Remove the NAT keep-alive job if present. if (m_natKeepAliveJobs.ContainsKey(bindings[index].RemoteSIPSocket)) { m_natKeepAliveJobs[bindings[index].RemoteSIPSocket].CancelJob(); } } } FireSIPMonitorLogEvent(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPRegistrarBindingRemoval, sipAccount.Owner, sipAccount.Id.ToString(), SIPURI.ParseSIPURIRelaxed(sipAccountAOR))); responseStatus = SIPResponseStatusCodesEnum.Ok; } else { // Remove all header cannot be present with other headers and must have an Expiry equal to 0. responseStatus = SIPResponseStatusCodesEnum.BadRequest; } #endregion } else { int requestedExpiry = (contactHeaderExpiresValue != -1) ? contactHeaderExpiresValue : expiresHeaderValue; requestedExpiry = (requestedExpiry == -1) ? maxAllowedExpiry : requestedExpiry; // This will happen if the Expires header and the Expiry on the Contact are both missing. bindingExpiry = (requestedExpiry > maxAllowedExpiry) ? maxAllowedExpiry : requestedExpiry; bindingExpiry = (bindingExpiry < MINIMUM_EXPIRY_SECONDS) ? MINIMUM_EXPIRY_SECONDS : bindingExpiry; bindingURI.Parameters.Remove(m_sipExpiresParameterKey); SIPRegistrarBinding binding = GetBindingForContactURI(bindings, bindingURI.ToString()); if (binding != null) { if (requestedExpiry <= 0) { FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingExpired, "Binding expired by client for " + sipAccountAOR + " from " + remoteSIPEndPoint.ToString() + ".", sipAccount.SIPUsername)); bindings.Remove(binding); m_bindingsPersistor.Delete(binding); bindingExpiry = 0; FireSIPMonitorLogEvent(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPRegistrarBindingRemoval, sipAccount.Owner, sipAccount.Id.ToString(), SIPURI.ParseSIPURIRelaxed(sipAccountAOR))); // Remove the NAT keep-alive job if present. if (m_natKeepAliveJobs.ContainsKey(binding.RemoteSIPSocket)) { m_natKeepAliveJobs[binding.RemoteSIPSocket].CancelJob(); } } else { FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Registrar, "Binding update request for " + sipAccountAOR + " from " + remoteSIPEndPoint.ToString() + ", expiry requested " + requestedExpiry + "s granted " + bindingExpiry + "s.", sipAccount.Owner)); binding.RefreshBinding(bindingExpiry, remoteSIPEndPoint, proxySIPEndPoint, registrarSIPEndPoint, sipAccount.DontMangleEnabled); DateTime startTime = DateTime.Now; m_bindingsPersistor.Update(binding); TimeSpan duration = DateTime.Now.Subtract(startTime); //FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.RegistrarTiming, "Binding database update time for " + sipAccountAOR + " took " + duration.TotalMilliseconds + "ms.", null)); FireSIPMonitorLogEvent(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPRegistrarBindingUpdate, sipAccount.Owner, sipAccount.Id.ToString(), SIPURI.ParseSIPURIRelaxed(sipAccountAOR))); // Add a NAT keep-alive job if required. if (sipAccount.SendNATKeepAlives && proxySIPEndPoint != null) { AddNATKeepAliveJob(sipAccount, remoteSIPEndPoint, proxySIPEndPoint, binding, bindingExpiry); } } } else { if (requestedExpiry > 0) { FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingInProgress, "New binding request for " + sipAccountAOR + " from " + remoteSIPEndPoint.ToString() + ", expiry requested " + requestedExpiry + "s granted " + bindingExpiry + "s.", sipAccount.Owner)); if (bindings.Count >= m_maxBindingsPerAccount) { // Need to remove the oldest binding to stay within limit. //SIPRegistrarBinding oldestBinding = m_bindingsPersistor.Get(b => b.SIPAccountId == sipAccount.Id, null, 0, Int32.MaxValue).OrderBy(x => x.LastUpdateUTC).Last(); SIPRegistrarBinding oldestBinding = bindings.OrderBy(x => x.LastUpdate).Last(); FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingInProgress, "Binding limit exceeded for " + sipAccountAOR + " from " + remoteSIPEndPoint.ToString() + " removing oldest binding to stay within limit of " + m_maxBindingsPerAccount + ".", sipAccount.Owner)); m_bindingsPersistor.Delete(oldestBinding); if (m_natKeepAliveJobs.ContainsKey(binding.RemoteSIPSocket)) { m_natKeepAliveJobs[binding.RemoteSIPSocket].CancelJob(); } } SIPRegistrarBinding newBinding = new SIPRegistrarBinding(sipAccount, bindingURI, callId, cseq, userAgent, remoteSIPEndPoint, proxySIPEndPoint, registrarSIPEndPoint, bindingExpiry); DateTime startTime = DateTime.Now; bindings.Add(newBinding); m_bindingsPersistor.Add(newBinding); TimeSpan duration = DateTime.Now.Subtract(startTime); //FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.RegistrarTiming, "Binding database add time for " + sipAccountAOR + " took " + duration.TotalMilliseconds + "ms.", null)); // Add a NAT keep-alive job if required. if (sipAccount.SendNATKeepAlives && proxySIPEndPoint != null) { AddNATKeepAliveJob(sipAccount, remoteSIPEndPoint, proxySIPEndPoint, newBinding, bindingExpiry); } FireSIPMonitorLogEvent(new SIPMonitorMachineEvent(SIPMonitorMachineEventTypesEnum.SIPRegistrarBindingUpdate, sipAccount.Owner, sipAccount.Id.ToString(), SIPURI.ParseSIPURIRelaxed(sipAccountAOR))); } else { FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.BindingFailed, "New binding received for " + sipAccountAOR + " with expired contact," + bindingURI.ToString() + " no update.", sipAccount.Owner)); bindingExpiry = 0; } } responseStatus = SIPResponseStatusCodesEnum.Ok; } } return(bindings); } catch (Exception excp) { logger.Error("Exception UpdateBinding. " + excp.Message); FireSIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Registrar, SIPMonitorEventTypesEnum.Error, "Registrar error updating binding: " + excp.Message + " Binding not updated.", sipAccount.SIPUsername)); responseStatus = SIPResponseStatusCodesEnum.InternalServerError; return(null); } }
/// <summary> /// Initiates a SUBSCRIBE request to a notification server. /// </summary> /// <param name="subscribeToURI">The SIP user that dialog notifications are being subscribed to.</param> public void Subscribe(SIPURI subscribeURI, int expiry, SIPEventPackage sipEventPackage, string subscribeCallID, SIPURI contactURI) { try { if (m_attempts >= MAX_SUBSCRIBE_ATTEMPTS) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.NotifierClient, SIPMonitorEventTypesEnum.SubscribeFailed, "Susbcription to " + subscribeURI.ToString() + " reached the maximum number of allowed attempts without a failure condition.", null)); m_subscribed = false; SubscriptionFailed(subscribeURI, SIPResponseStatusCodesEnum.InternalServerError, "Subscription reached the maximum number of allowed attempts."); m_waitForSubscribeResponse.Set(); } else { m_attempts++; m_localCSeq++; SIPRequest subscribeRequest = m_sipTransport.GetRequest( SIPMethodsEnum.SUBSCRIBE, m_resourceURI, new SIPToHeader(null, subscribeURI, m_subscriptionToTag), null); if (contactURI != null) { subscribeRequest.Header.Contact = new List <SIPContactHeader>() { new SIPContactHeader(null, contactURI) }; } subscribeRequest.Header.From = new SIPFromHeader(null, new SIPURI(m_authUsername, m_authDomain, null, SIPSchemesEnum.sip, SIPProtocolsEnum.udp), m_subscriptionFromTag); subscribeRequest.Header.CSeq = m_localCSeq; subscribeRequest.Header.Expires = expiry; subscribeRequest.Header.Event = sipEventPackage.ToString(); subscribeRequest.Header.CallId = subscribeCallID; if (!m_filter.IsNullOrBlank()) { subscribeRequest.Body = m_filter; subscribeRequest.Header.ContentLength = m_filter.Length; subscribeRequest.Header.ContentType = m_filterTextType; } SIPEndPoint dstEndPoint = m_outboundProxy; if (dstEndPoint == null) { SIPDNSLookupResult lookupResult = m_sipTransport.GetURIEndPoint(m_resourceURI, false); if (lookupResult.LookupError != null) { Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.NotifierClient, SIPMonitorEventTypesEnum.SubscribeFailed, "Could not resolve " + m_resourceURI.Host + ", " + lookupResult.LookupError, null)); } else { dstEndPoint = lookupResult.GetSIPEndPoint(); } } SIPNonInviteTransaction subscribeTransaction = m_sipTransport.CreateNonInviteTransaction(subscribeRequest, dstEndPoint, null, m_outboundProxy); subscribeTransaction.NonInviteTransactionFinalResponseReceived += SubscribeTransactionFinalResponseReceived; subscribeTransaction.NonInviteTransactionTimedOut += SubsribeTransactionTimedOut; m_sipTransport.SendSIPReliable(subscribeTransaction); LastSubscribeAttempt = DateTime.Now; } } catch (Exception excp) { logger.LogError("Exception SIPNotifierClient Subscribe. " + excp.Message); SubscriptionFailed(m_resourceURI, SIPResponseStatusCodesEnum.InternalServerError, "Exception Subscribing. " + excp.Message); m_waitForSubscribeResponse.Set(); } }
static void Main(string[] args) { Console.WriteLine("SIPSorcery client user agent example."); Console.WriteLine("Press ctrl-c to exit."); // Plumbing code to facilitate a graceful exit. CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream. bool isCallHungup = false; bool hasCallFailed = false; AddConsoleLogger(); SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI); if (args != null && args.Length > 0) { if (!SIPURI.TryParse(args[0])) { Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}"); } else { callUri = SIPURI.ParseSIPURIRelaxed(args[0]); } } Log.LogInformation($"Call destination {callUri}."); var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false); Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}."); var dstAddress = lookupResult.GetSIPEndPoint().Address; // Set up a default SIP transport. var sipTransport = new SIPTransport(); var listenAddress = dstAddress.AddressFamily == AddressFamily.InterNetwork ? IPAddress.Any : IPAddress.IPv6Any; if (callUri.Scheme == SIPSchemesEnum.sip && callUri.Protocol == SIPProtocolsEnum.udp) { sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, 0))); } else if (callUri.Scheme == SIPSchemesEnum.sip && callUri.Protocol == SIPProtocolsEnum.tcp) { sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, 0))); } else if (callUri.Scheme == SIPSchemesEnum.sips || callUri.Protocol == SIPProtocolsEnum.tls) { sipTransport.AddSIPChannel(new SIPTLSChannel(new X509Certificate2("localhost.pfx"), new IPEndPoint(listenAddress, 0))); } EnableTraceLogs(sipTransport); IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress); // Initialise an RTP session to receive the RTP packets from the remote SIP server. Socket rtpSocket = null; Socket controlSocket = null; NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out rtpSocket, out controlSocket); var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null); var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null); // Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call. var uac = new SIPClientUserAgent(sipTransport); uac.CallTrying += (uac, resp) => { Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}."); }; uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}."); uac.CallFailed += (uac, err) => { Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}"); hasCallFailed = true; }; uac.CallAnswered += (uac, resp) => { if (resp.Status == SIPResponseStatusCodesEnum.Ok) { Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); // Only set the remote RTP end point if there hasn't already been a packet received on it. if (_remoteRtpEndPoint == null) { _remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body); Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}."); } } else { Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}."); } }; // The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call. sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) => { if (sipRequest.Method == SIPMethodsEnum.BYE) { SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null); SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); byeTransaction.SendFinalResponse(byeResponse); if (uac.IsUACAnswered) { Log.LogInformation("Call was hungup by remote server."); isCallHungup = true; rtpCts.Cancel(); } } }; // It's a good idea to start the RTP receiving socket before the call request is sent. // A SIP server will generally start sending RTP as soon as it has processed the incoming call request and // being ready to receive will stop any ICMP error response being generated. Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts)); Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts)); // Start the thread that places the call. SIPCallDescriptor callDescriptor = new SIPCallDescriptor( SIPConstants.SIP_DEFAULT_USERNAME, null, callUri.ToString(), SIPConstants.SIP_DEFAULT_FROMURI, callUri.CanonicalAddress, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(), null); uac.Call(callDescriptor); // Ctrl-c will gracefully exit the call at any point. Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e) { e.Cancel = true; rtpCts.Cancel(); }; // At this point the call has been initiated and everything will be handled in an event handler or on the RTP // receive task. The code below is to gracefully exit. // Wait for a signal saying the call failed, was cancelled with ctrl-c or completed. rtpCts.Token.WaitHandle.WaitOne(); Log.LogInformation("Exiting..."); rtpSocket?.Close(); controlSocket?.Close(); if (!isCallHungup && uac != null) { if (uac.IsUACAnswered) { Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}."); uac.Hangup(); } else if (!hasCallFailed) { Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}."); uac.Cancel(); } // Give the BYE or CANCEL request time to be transmitted. Log.LogInformation("Waiting 1s for call to clean up..."); Task.Delay(1000).Wait(); } SIPSorcery.Net.DNSManager.Stop(); if (sipTransport != null) { Log.LogInformation("Shutting down SIP transport..."); sipTransport.Shutdown(); } }
public static SIPDNSLookupResult ResolveSIPService(SIPURI sipURI, bool async) { try { if (sipURI == null) { throw new ArgumentNullException("sipURI", "Cannot resolve SIP service on a null URI."); } string host = sipURI.Host; int port = (sipURI.Scheme == SIPSchemesEnum.sip) ? m_defaultSIPPort : m_defaultSIPSPort; bool explicitPort = false; if (sipURI.Host.IndexOf(':') != -1) { host = sipURI.Host.Split(':')[0]; Int32.TryParse(sipURI.Host.Split(':')[1], out port); explicitPort = true; } if (Regex.Match(host, @"^\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}$").Success) { // Target is an IP address, no DNS lookup required. IPAddress hostIP = IPAddress.Parse(host); SIPDNSLookupEndPoint sipLookupEndPoint = new SIPDNSLookupEndPoint(new SIPEndPoint(sipURI.Protocol, new IPEndPoint(hostIP, port)), 0); SIPDNSLookupResult result = new SIPDNSLookupResult(sipURI); result.AddLookupResult(sipLookupEndPoint); return result; } else if (explicitPort) { // Target is a hostname with an explicit port, DNS lookup for A or AAAA record. return DNSARecordLookup(host, port, async, sipURI); } else { // Target is a hostname with no explicit port, use the whole NAPTR->SRV->A lookup procedure. SIPDNSLookupResult sipLookupResult = new SIPDNSLookupResult(sipURI); // Do without the NAPTR lookup for the time being. Very few organisations appear to use them and it can cost up to 2.5s to get a failed resolution. /*SIPMonitorLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Unknown, SIPMonitorEventTypesEnum.DNS, "SIP DNS full lookup requested for " + sipURI.ToString() + ".", null)); DNSNAPTRRecordLookup(host, async, ref sipLookupResult); if (sipLookupResult.Pending) { if (!m_inProgressSIPServiceLookups.Contains(sipURI.ToString())) { m_inProgressSIPServiceLookups.Add(sipURI.ToString()); ThreadPool.QueueUserWorkItem(delegate { ResolveSIPService(sipURI, false); }); } return sipLookupResult; }*/ DNSSRVRecordLookup(sipURI.Scheme, sipURI.Protocol, host, async, ref sipLookupResult); if (sipLookupResult.Pending) { //logger.Debug("SIPDNSManager SRV lookup for " + host + " is pending."); return sipLookupResult; } else { //logger.Debug("SIPDNSManager SRV lookup for " + host + " is final."); SIPDNSServiceResult nextSRVRecord = sipLookupResult.GetNextUnusedSRV(); int lookupPort = (nextSRVRecord != null) ? nextSRVRecord.Port : port; return DNSARecordLookup(nextSRVRecord, host, lookupPort, async, sipLookupResult.URI); } } } catch (Exception excp) { logger.Error("Exception SIPDNSManager ResolveSIPService (" + sipURI.ToString() + "). " + excp.Message); m_inProgressSIPServiceLookups.Remove(sipURI.ToString()); return new SIPDNSLookupResult(sipURI, excp.Message); } }
private void Subscribe(SIPTransaction subscribeTransaction) { try { SIPRequest sipRequest = subscribeTransaction.TransactionRequest; string fromUser = sipRequest.Header.From.FromURI.User; string fromHost = sipRequest.Header.From.FromURI.Host; string canonicalDomain = GetCanonicalDomain_External(fromHost, true); if (canonicalDomain == null) { FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Subscribe request for " + fromHost + " rejected as no matching domain found.", null)); SIPResponse noDomainResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Forbidden, "Domain not serviced"); subscribeTransaction.SendFinalResponse(noDomainResponse); return; } SIPAccountAsset sipAccountAsset = m_sipAssetPersistor.Get(s => s.SIPUsername == fromUser && s.SIPDomain == canonicalDomain); SIPRequestAuthenticationResult authenticationResult = SIPRequestAuthenticator_External(subscribeTransaction.LocalSIPEndPoint, subscribeTransaction.RemoteEndPoint, sipRequest, sipAccountAsset.SIPAccount, FireProxyLogEvent); if (!authenticationResult.Authenticated) { // 401 Response with a fresh nonce needs to be sent. SIPResponse authReqdResponse = SIPTransport.GetResponse(sipRequest, authenticationResult.ErrorResponse, null); authReqdResponse.Header.AuthenticationHeader = authenticationResult.AuthenticationRequiredHeader; subscribeTransaction.SendFinalResponse(authReqdResponse); if (authenticationResult.ErrorResponse == SIPResponseStatusCodesEnum.Forbidden) { FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Warn, "Forbidden " + fromUser + "@" + canonicalDomain + " does not exist, " + sipRequest.Header.ProxyReceivedFrom.ToString() + ", " + sipRequest.Header.UserAgent + ".", null)); } else { FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Authentication required for " + fromUser + "@" + canonicalDomain + " from " + subscribeTransaction.RemoteEndPoint + ".", sipAccountAsset.Owner)); } return; } else { if (sipRequest.Header.To.ToTag != null) { // Request is to renew an existing subscription. SIPResponseStatusCodesEnum errorResponse = SIPResponseStatusCodesEnum.None; string errorResponseReason = null; string sessionID = m_subscriptionsManager.RenewSubscription(sipRequest, out errorResponse, out errorResponseReason); if (errorResponse != SIPResponseStatusCodesEnum.None) { // A subscription renewal attempt failed SIPResponse renewalErrorResponse = SIPTransport.GetResponse(sipRequest, errorResponse, errorResponseReason); subscribeTransaction.SendFinalResponse(renewalErrorResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription renewal failed for event type " + sipRequest.Header.Event + " " + sipRequest.URI.ToString() + ", " + errorResponse + " " + errorResponseReason + ".", sipAccountAsset.Owner)); } else if (sipRequest.Header.Expires == 0) { // Existing subscription was closed. SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); subscribeTransaction.SendFinalResponse(okResponse); } else { // Existing subscription was found. SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); subscribeTransaction.SendFinalResponse(okResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeRenew, "Subscription renewal for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + " and expiry " + sipRequest.Header.Expires + ".", sipAccountAsset.Owner)); m_subscriptionsManager.SendFullStateNotify(sessionID); } } else { // Authenticated but the this is a new subscription request and authorisation to subscribe to the requested resource also needs to be checked. SIPURI canonicalResourceURI = sipRequest.URI.CopyOf(); string resourceCanonicalDomain = GetCanonicalDomain_External(canonicalResourceURI.Host, true); canonicalResourceURI.Host = resourceCanonicalDomain; SIPAccountAsset resourceSIPAccount = null; if (resourceCanonicalDomain == null) { SIPResponse notFoundResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain " + resourceCanonicalDomain + " not serviced"); subscribeTransaction.SendFinalResponse(notFoundResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", domain not serviced.", sipAccountAsset.Owner)); return; } if (canonicalResourceURI.User != m_wildcardUser) { resourceSIPAccount = m_sipAssetPersistor.Get(s => s.SIPUsername == canonicalResourceURI.User && s.SIPDomain == canonicalResourceURI.Host); if (resourceSIPAccount == null) { SIPResponse notFoundResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Requested resource does not exist"); subscribeTransaction.SendFinalResponse(notFoundResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", SIP account does not exist.", sipAccountAsset.Owner)); return; } } // Check the owner permissions on the requesting and subscribed resources. bool authorised = false; string adminID = null; if (canonicalResourceURI.User == m_wildcardUser || sipAccountAsset.Owner == resourceSIPAccount.Owner) { authorised = true; FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Subscription to " + canonicalResourceURI.ToString() + " authorised due to common owner.", sipAccountAsset.Owner)); } else { // Lookup the customer record for the requestor and check the administrative level on it. Customer requestingCustomer = GetCustomer_External(c => c.CustomerUsername == sipAccountAsset.Owner); adminID = requestingCustomer.AdminId; if (!resourceSIPAccount.AdminMemberId.IsNullOrBlank() && requestingCustomer.AdminId == resourceSIPAccount.AdminMemberId) { authorised = true; FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Subscription to " + canonicalResourceURI.ToString() + " authorised due to requestor admin permissions for domain " + resourceSIPAccount.AdminMemberId + ".", sipAccountAsset.Owner)); } else if (requestingCustomer.AdminId == m_topLevelAdminID) { authorised = true; FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAuth, "Subscription to " + canonicalResourceURI.ToString() + " authorised due to requestor having top level admin permissions.", sipAccountAsset.Owner)); } } if (authorised) { // Request is to create a new subscription. SIPResponseStatusCodesEnum errorResponse = SIPResponseStatusCodesEnum.None; string errorResponseReason = null; string toTag = CallProperties.CreateNewTag(); string sessionID = m_subscriptionsManager.SubscribeClient(sipAccountAsset.Owner, adminID, sipRequest, toTag, canonicalResourceURI, out errorResponse, out errorResponseReason); if (errorResponse != SIPResponseStatusCodesEnum.None) { SIPResponse subscribeErrorResponse = SIPTransport.GetResponse(sipRequest, errorResponse, errorResponseReason); subscribeTransaction.SendFinalResponse(subscribeErrorResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAccept, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", " + errorResponse + " " + errorResponseReason + ".", sipAccountAsset.Owner)); } else { SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null); okResponse.Header.To.ToTag = toTag; okResponse.Header.Expires = sipRequest.Header.Expires; okResponse.Header.Contact = new List <SIPContactHeader>() { new SIPContactHeader(null, new SIPURI(SIPSchemesEnum.sip, subscribeTransaction.LocalSIPEndPoint)) }; subscribeTransaction.SendFinalResponse(okResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeAccept, "Subscription accepted for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + " and expiry " + sipRequest.Header.Expires + ".", sipAccountAsset.Owner)); if (sessionID != null) { m_subscriptionsManager.SendFullStateNotify(sessionID); } } } else { SIPResponse forbiddenResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Forbidden, "Requested resource not authorised"); subscribeTransaction.SendFinalResponse(forbiddenResponse); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.SubscribeFailed, "Subscription failed for " + sipRequest.URI.ToString() + ", event type " + sipRequest.Header.Event + ", requesting account " + sipAccountAsset.Owner + " was not authorised.", sipAccountAsset.Owner)); } } } } catch (Exception excp) { logger.Error("Exception notifiercore subscribing. " + excp.Message + "\r\n" + subscribeTransaction.TransactionRequest.ToString()); FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.Notifier, SIPMonitorEventTypesEnum.Error, "Exception notifiercore subscribing. " + excp.Message, null)); SIPResponse errorResponse = SIPTransport.GetResponse(subscribeTransaction.TransactionRequest, SIPResponseStatusCodesEnum.InternalServerError, null); subscribeTransaction.SendFinalResponse(errorResponse); } }