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);
}
}
