private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Console.WriteLine("INVITE received from " + localSIPEndPoint.ToString());
IPEndPoint sipPhoneEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPServerUserAgent uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
uasTransaction.SendInformationalResponse(tryingResponse);
if (m_xmppClient == null)
{
m_xmppClient = new XMPPClient(XMPP_SERVER, XMPP_SERVER_PORT, XMPP_REALM, m_xmppUsername, m_xmppPassword);
m_xmppClient.Disconnected += XMPPDisconnected;
m_xmppClient.IsBound += () => { XMPPPlaceCall(uas); };
ThreadPool.QueueUserWorkItem(delegate { m_xmppClient.Connect(); });
}
else
{
XMPPPlaceCall(uas);
}
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
Console.WriteLine("Matching CANCEL request received " + sipRequest.URI.ToString() + ".");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
Console.WriteLine("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Console.WriteLine("BYE request received.");
if (m_activeCalls.ContainsKey(sipRequest.Header.CallId))
{
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
m_activeCalls[sipRequest.Header.CallId].TerminateXMPPCall();
m_activeCalls.Remove(sipRequest.Header.CallId);
}
else
{
SIPResponse doesntExistResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(doesntExistResponse);
}
}
}
public async Task CancelCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport aliceTransport = new SIPTransport();
aliceTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var alice = new SIPUserAgent(aliceTransport, null, true);
SIPServerUserAgent uas = null;
// Auto accept but NOT answering.
alice.OnIncomingCall += (ua, req) => uas = ua.AcceptCall(req);
SIPTransport bobTransport = new SIPTransport();
bobTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var bob = new SIPUserAgent(bobTransport, null, true);
var callTask = bob.Call(alice.ContactURI.ToString(), null, null, CreateMediaSession());
await Task.Delay(500);
Assert.True(bob.IsRinging);
Assert.NotNull(uas);
Assert.False(uas.IsCancelled);
bob.Cancel();
await Task.Delay(500);
Assert.False(alice.IsCallActive);
Assert.False(bob.IsCallActive);
Assert.True(uas.IsCancelled);
}
/// <summary>
/// Answers an incoming SIP call.
/// </summary>
public async Task <bool> Answer()
{
if (m_pendingIncomingCall == null)
{
StatusMessage(this, $"There was no pending call available to answer.");
return(false);
}
else
{
var sipRequest = m_pendingIncomingCall.ClientTransaction.TransactionRequest;
// Assume that if the INVITE request does not contain an SDP offer that it will be an
// audio only call.
bool hasAudio = true;
bool hasVideo = false;
if (sipRequest.Body != null)
{
SDP offerSDP = SDP.ParseSDPDescription(sipRequest.Body);
hasAudio = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
hasVideo = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.video && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
}
MediaSession = CreateMediaSession();
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
bool result = await m_userAgent.Answer(m_pendingIncomingCall, MediaSession);
m_pendingIncomingCall = null;
return(result);
}
}
private static void XMPPPlaceCall(SIPServerUserAgent uas)
{
if (!uas.IsCancelled)
{
XMPPPhoneSession phoneSession = m_xmppClient.GetPhoneSession();
SIPToXMPPCall call = new SIPToXMPPCall(uas, phoneSession, m_sipTransport, m_ipAddress);
m_activeCalls.Add(uas.CallRequest.Header.CallId, call);
call.Call(uas.CallRequest.URI.User);
}
}
public async Task BlindTransferCancelUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 343" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF +
"Min-SE: 90" + m_CRLF +
"" + m_CRLF +
"v=0" + m_CRLF +
"o=- 3785527268 3785527269 IN IP4 192.168.11.50" + m_CRLF +
"s=pjmedia" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 4032 RTP/AVP 0 101" + m_CRLF +
"c=IN IP4 192.168.11.50" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
SIPServerUserAgent mockUas = new SIPServerUserAgent(transport, null, null, null, SIPCallDirection.In, null, null, null, uasTx);
await userAgent.Answer(mockUas, CreateMediaSession());
CancellationTokenSource cts = new CancellationTokenSource();
var blindTransferTask = userAgent.BlindTransfer(SIPURI.ParseSIPURIRelaxed("127.0.0.1"), TimeSpan.FromSeconds(2), cts.Token);
cts.Cancel();
Assert.False(await blindTransferTask);
//await Assert.ThrowsAnyAsync<TaskCanceledException>(async () => { bool result = ; });
}
/// <summary>
/// Answers an incoming SIP call.
/// </summary>
public async Task <bool> Answer()
{
if (m_pendingIncomingCall == null)
{
StatusMessage(this, $"There was no pending call available to answer.");
return(false);
}
else
{
var sipRequest = m_pendingIncomingCall.ClientTransaction.TransactionRequest;
SDP offerSDP = SDP.ParseSDPDescription(sipRequest.Body);
bool hasAudio = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
bool hasVideo = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.video && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
AudioOptions audioOpts = new AudioOptions {
AudioSource = AudioSourcesEnum.None
};
if (hasAudio)
{
audioOpts = new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
OutputDeviceIndex = m_audioOutDeviceIndex,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
};
}
VideoOptions videoOpts = new VideoOptions {
VideoSource = VideoSourcesEnum.None
};
if (hasVideo)
{
videoOpts = new VideoOptions
{
VideoSource = VideoSourcesEnum.TestPattern,
SourceFile = RtpAVSession.VIDEO_TESTPATTERN,
SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND
};
}
MediaSession = new RtpAVSession(audioOpts, videoOpts);
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
bool result = await m_userAgent.Answer(m_pendingIncomingCall, MediaSession);
m_pendingIncomingCall = null;
return(result);
}
}
public SIPToXMPPCall(SIPServerUserAgent uas, XMPPPhoneSession xmppCall, SIPTransport sipTransport, IPAddress ipAddress)
{
m_uas = uas;
m_xmppCall = xmppCall;
m_sipTransport = sipTransport;
m_ipAddress = ipAddress;
m_uas.CallCancelled += SIPCallCancelled;
m_xmppCall.Accepted += Answered;
m_xmppCall.Rejected += CallFailed;
m_xmppCall.Hungup += Hangup;
}
/// <summary>
/// Cleans up after a SIP call has completely finished.
/// </summary>
private void CallFinished()
{
if (_mediaManager != null)
{
_mediaManager.EndCall();
_mediaManager = null;
}
m_uac = null;
m_uas = null;
CallEnded();
}
private DialPlanLineContext GetDummyDialPlanContext(string testDialPlan, string dst)
{
SIPDialPlan dialPlan = new SIPDialPlan(null, null, null, testDialPlan, SIPDialPlanScriptTypesEnum.Asterisk);
SIPTransactionEngine transactionEngine = new SIPTransactionEngine();
SIPTransport sipTransport = new SIPTransport(MockSIPDNSManager.Resolve, transactionEngine);
SIPURI dummyURI = SIPURI.ParseSIPURI(dst);
SIPRequest inviteRequest = GetDummyINVITERequest(dummyURI);
SIPEndPoint dummyEndPoint = SIPEndPoint.ParseSIPEndPoint("udp:0.0.0.0:5060");
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(inviteRequest, dummyEndPoint, dummyEndPoint, null);
SIPServerUserAgent uas = new SIPServerUserAgent(sipTransport, null, "test", "sipsorcery.com", SIPCallDirection.In, null, null, null, uasTransaction);
DialPlanLineContext dialPlanContext = new DialPlanLineContext(null, null, null, null, uas, dialPlan, null, null, null, null);
return(dialPlanContext);
}
/// <summary>
/// Cleans up after a SIP call has completely finished.
/// </summary>
private void CallFinished()
{
if (_mediaManager != null)
{
_mediaManager.EndCall();
_mediaManager = null;
}
//_cancelCallTokenSource.Cancel();
m_uac = null;
m_uas = null;
CallEnded();
}
/// <summary>
/// Answers an incoming SIP call.
/// </summary>
public async Task Answer()
{
if (m_pendingIncomingCall == null)
{
StatusMessage(this, $"There was no pending call available to answer.");
}
else
{
var sipRequest = m_pendingIncomingCall.ClientTransaction.TransactionRequest;
SDP offerSDP = SDP.ParseSDPDescription(sipRequest.Body);
bool hasAudio = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.audio);
bool hasVideo = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.video);
AudioOptions audioOpts = new AudioOptions {
AudioSource = AudioSourcesEnum.None
};
if (hasAudio)
{
audioOpts = new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
};
}
VideoOptions videoOpts = new VideoOptions {
VideoSource = VideoSourcesEnum.None
};
if (hasVideo)
{
videoOpts = new VideoOptions
{
VideoSource = VideoSourcesEnum.TestPattern,
SourceFile = RtpAVSession.VIDEO_TESTPATTERN,
SourceFramesPerSecond = VIDEO_LIVE_FRAMES_PER_SECOND
};
}
MediaSession = new RtpAVSession(sipRequest.RemoteSIPEndPoint.Address.AddressFamily, audioOpts, videoOpts);
m_userAgent.RemotePutOnHold += OnRemotePutOnHold;
m_userAgent.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.Answer(m_pendingIncomingCall, MediaSession);
m_pendingIncomingCall = null;
}
}
/// <summary>
/// Hangs up the current call.
/// </summary>
/// <param name="uas">The user agent server to hangup the call on.</param>
private static async Task Hangup(SIPServerUserAgent uas)
{
try
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to hangup...");
await Task.Delay(1000);
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Hangup. {excp.Message}");
}
}
/// <summary>
/// Answers an incoming SIP call.
/// </summary>
public async Task Answer()
{
if (m_pendingIncomingCall == null)
{
StatusMessage(this, $"There was no pending call available to answer.");
}
else
{
var sipRequest = m_pendingIncomingCall.ClientTransaction.TransactionRequest;
m_rtpMediaSessionManager.Create(sipRequest.RemoteSIPEndPoint.Address.AddressFamily);
m_rtpMediaSessionManager.RTPMediaSession.RemotePutOnHold += OnRemotePutOnHold;
m_rtpMediaSessionManager.RTPMediaSession.RemoteTookOffHold += OnRemoteTookOffHold;
await m_userAgent.Answer(m_pendingIncomingCall, m_rtpMediaSessionManager.RTPMediaSession);
m_pendingIncomingCall = null;
}
}
/// <param name="rtpListenAddress">The local IP address to establish the RTP listener socket on.</param>
/// <param name="sdpAdvertiseAddress">The public IP address to put into the SDP sent back to the caller.</param>
/// <param name="request">The INVITE request that instigated the RTP diagnostics job.</param>
public RTPDiagnosticsJob(IPAddress rtpListenAddress, IPAddress sdpAdvertiseAddress, SIPServerUserAgent uas, SIPRequest request)
{
m_request = request;
m_remoteSDP = SDP.ParseSDPDescription(request.Body);
RemoteRTPEndPoint = new IPEndPoint(IPAddress.Parse(m_remoteSDP.Connection.ConnectionAddress), m_remoteSDP.Media[0].Port);
UAS = uas;
//m_rawSourceStream = new RawSourceWaveStream(m_outStream, WaveFormat.CreateMuLawFormat(8000, 1));
//m_waveFileWriter = new WaveFileWriter("out.wav", new WaveFormat(8000, 16, 1));
m_waveFileWriter = new WaveFileWriter("out.wav", new WaveFormat(8000, 16, 1));
//m_outPCMStream = WaveFormatConversionStream.CreatePcmStream(m_rawSourceStream);
//m_rawRTPPayloadWriter = new StreamWriter("out.rtp");
//m_rawRTPPayloadReader = new StreamReader("in.rtp");
//IPEndPoint rtpListenEndPoint = null;
IPEndPoint rtpListenEndPoint = null;
NetServices.CreateRandomUDPListener(rtpListenAddress, RTP_PORTRANGE_START, RTP_PORTRANGE_END, m_inUsePorts, out rtpListenEndPoint);
RTPListenEndPoint = rtpListenEndPoint;
m_inUsePorts.Add(rtpListenEndPoint.Port);
//RTPListenEndPoint = new IPEndPoint(rtpListenAddress, RTP_PORTRANGE_START);
m_rtpChannel = new RTPChannel(RTPListenEndPoint);
m_rtpChannel.SampleReceived += SampleReceived;
ThreadPool.QueueUserWorkItem(delegate { GetAudioSamples(); });
LocalSDP = new SDP()
{
SessionId = Crypto.GetRandomString(6),
Address = sdpAdvertiseAddress.ToString(),
SessionName = "sipsorcery",
Timing = "0 0",
Connection = new SDPConnectionInformation(sdpAdvertiseAddress.ToString()),
Media = new List <SDPMediaAnnouncement>()
{
new SDPMediaAnnouncement()
{
Media = SDPMediaTypesEnum.audio,
Port = RTPListenEndPoint.Port,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU)
}
}
}
};
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2("localhost.pfx");
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
RtpAVSession rtpSession = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)SDPMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
rtpSession = new RtpAVSession(
new AudioOptions {
AudioSource = AudioSourcesEnum.Music, SourceFile = executableDir + "/" + AUDIO_FILE_PCMU
}, null);
rtpSession.setRemoteDescription(new RTCSessionDescription {
type = RTCSdpType.offer, sdp = offerSdp
});
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
await sipTransport.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
rtpCts = new CancellationTokenSource();
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.CloseSession(null);
};
rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = await rtpSession.createAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
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, 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();
}
}
public async Task HangupUserAgentUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport(false, MockSIPDNSManager.Resolve);
MockSIPChannel mockChannel = new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0));
transport.AddSIPChannel(mockChannel);
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 343" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF +
"Min-SE: 90" + m_CRLF +
"" + m_CRLF +
"v=0" + m_CRLF +
"o=- 3785527268 3785527269 IN IP4 192.168.11.50" + m_CRLF +
"s=pjmedia" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 4032 RTP/AVP 0 101" + m_CRLF +
"c=IN IP4 192.168.11.50" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
SIPServerUserAgent mockUas = new SIPServerUserAgent(transport, null, null, null, SIPCallDirection.In, null, null, null, uasTx);
await userAgent.Answer(mockUas, CreateMediaSession());
// Incremented Cseq and modified Via header from original request. Means the request is the same dialog but different tx.
string inviteReqStr2 = "BYE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3700" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3533 BYE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"";
mockChannel.FireMessageReceived(dummySipEndPoint, dummySipEndPoint, Encoding.UTF8.GetBytes(inviteReqStr2));
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.ContactHost = Dns.GetHostName();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//if (File.Exists("localhost.pfx"))
//{
// var certificate = new X509Certificate2(@"localhost.pfx", "");
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.Any, SIPS_LISTEN_PORT)));
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.IPv6Any, SIPS_LISTEN_PORT)));
//}
//EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
// In practice there could be a number of reasons to reject the call. Unsupported extensions, mo matching codecs etc. etc.
if (sipRequest.Header.HasUnknownRequireExtension)
{
// The caller requires an extension that we don't support.
SIPSorcery.Sys.Log.Logger.LogWarning($"Rejecting incoming call due to one or more required exensions not being supported, required extensions: {sipRequest.Header.Require}.");
uas.Reject(SIPResponseStatusCodesEnum.NotImplemented, "Unsupported Require Extension", null);
}
else
{
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Simulating answer delay to test provisional response retransmits.
await Task.Delay(2000);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localSIPEndPoint.Address, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, rtpCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
try
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{localSIPEndPoint.ToString()}<-{remoteEndPoint.ToString()}: {sipRequest.StatusLine}");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
catch (Exception optionsExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Failed to send SIP OPTIONS response. {optionsExcp.Message}");
}
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpCts?.Cancel();
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to hangup...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
/// <summary>
/// Cleans up after a SIP call has completely finished.
/// </summary>
private void CallFinished(SIPDialogue dialogue)
{
m_pendingIncomingCall = null;
CallEnded(this);
}
public void GotRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
// Used in the proxy monitor messages only, plays no part in request routing.
string fromUser = (sipRequest.Header.From != null) ? sipRequest.Header.From.FromURI.User : null;
string fromURIStr = (sipRequest.Header.From != null) ? sipRequest.Header.From.FromURI.ToString() : "null";
//string toUser = (sipRequest.Header.To != null) ? sipRequest.Header.To.ToURI.User : null;
//string summaryStr = "req " + sipRequest.Method + " from=" + fromUser + ", to=" + toUser + ", " + remoteEndPoint.ToString();
//logger.Debug("AppServerCore GotRequest " + sipRequest.Method + " from " + remoteEndPoint.ToString() + " callid=" + sipRequest.Header.CallId + ".");
SIPDialogue dialogue = null;
// Check dialogue requests for an existing dialogue.
if ((sipRequest.Method == SIPMethodsEnum.BYE || sipRequest.Method == SIPMethodsEnum.INFO || sipRequest.Method == SIPMethodsEnum.INVITE ||
sipRequest.Method == SIPMethodsEnum.MESSAGE || sipRequest.Method == SIPMethodsEnum.NOTIFY || sipRequest.Method == SIPMethodsEnum.OPTIONS ||
sipRequest.Method == SIPMethodsEnum.REFER) &&
sipRequest.Header.From != null && sipRequest.Header.From.FromTag != null && sipRequest.Header.To != null && sipRequest.Header.To.ToTag != null)
{
dialogue = m_sipDialogueManager.GetDialogue(sipRequest);
}
if (dialogue != null && sipRequest.Method != SIPMethodsEnum.ACK)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Matching dialogue found for " + sipRequest.Method + " to " + sipRequest.URI.ToString() + " from " + remoteEndPoint + ".", dialogue.Owner));
if (sipRequest.Method != SIPMethodsEnum.REFER)
{
m_sipDialogueManager.ProcessInDialogueRequest(localSIPEndPoint, remoteEndPoint, sipRequest, dialogue);
}
else
{
m_sipDialogueManager.ProcessInDialogueReferRequest(localSIPEndPoint, remoteEndPoint, sipRequest, dialogue, m_callManager.BlindTransfer);
}
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
#region CANCEL request handling.
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Cancelling call for " + sipRequest.URI.ToString() + ".", fromUser));
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".", fromUser));
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
#endregion
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No dialogue matched for BYE to " + sipRequest.URI.ToString() + ".", fromUser));
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.REFER)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No dialogue matched for REFER to " + sipRequest.URI.ToString() + ".", fromUser));
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.ACK)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "No transaction matched for ACK for " + sipRequest.URI.ToString() + ".", fromUser));
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
#region INVITE request processing.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "AppServerCore INVITE received, uri=" + sipRequest.URI.ToString() + ", cseq=" + sipRequest.Header.CSeq + ".", null));
if (sipRequest.URI.User == m_dispatcherUsername)
{
// Incoming call from monitoring process checking the application server is still running.
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy, true);
//uasTransaction.CDR = null;
SIPServerUserAgent incomingCall = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, sipRequest.URI.User, sipRequest.URI.Host, SIPCallDirection.In, null, null, null, uasTransaction);
//incomingCall.NoCDR();
uasTransaction.NewCallReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(incomingCall); };
uasTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else if (GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false) != null)
{
// Call identified as outgoing call for application server serviced domain.
string fromDomain = GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPServerUserAgent outgoingCall = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, fromUser, fromDomain, SIPCallDirection.Out, GetSIPAccount_External, SIPRequestAuthenticator.AuthenticateSIPRequest, FireProxyLogEvent, uasTransaction);
uasTransaction.NewCallReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(outgoingCall); };
uasTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else if (GetCanonicalDomain_External(sipRequest.URI.Host, true) != null)
{
// Call identified as incoming call for application server serviced domain.
if (sipRequest.URI.User.IsNullOrBlank())
{
// Cannot process incoming call if no user is specified.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "INVITE received with an empty URI user " + sipRequest.URI.ToString() + ", returning address incomplete.", null));
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPResponse addressIncompleteResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.AddressIncomplete, "No user specified");
uasTransaction.SendFinalResponse(addressIncompleteResponse);
}
else
{
// Send the incoming call to the call manager for processing.
string uriUser = sipRequest.URI.User;
string uriDomain = GetCanonicalDomain_External(sipRequest.URI.Host, true);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPServerUserAgent incomingCall = new SIPServerUserAgent(m_sipTransport, m_outboundProxy, uriUser, uriDomain, SIPCallDirection.In, GetSIPAccount_External, null, FireProxyLogEvent, uasTransaction);
uasTransaction.NewCallReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(incomingCall); };
uasTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
}
else
{
// Return not found for non-serviced domain.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Domain not serviced " + sipRequest.URI.ToString() + ", returning not found.", null));
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPResponse notServicedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain not serviced");
uasTransaction.SendFinalResponse(notServicedResponse);
}
#endregion
}
else if (sipRequest.Method == SIPMethodsEnum.MESSAGE)
{
#region Processing non-INVITE requests that are accepted by the dialplan processing engine.
if (GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false) != null)
{
// Call identified as outgoing request for application server serviced domain.
string fromDomain = GetCanonicalDomain_External(sipRequest.Header.From.FromUserField.URI.Host, false);
SIPNonInviteTransaction nonInviteTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPNonInviteServerUserAgent outgoingRequest = new SIPNonInviteServerUserAgent(m_sipTransport, m_outboundProxy, fromUser, fromDomain, SIPCallDirection.Out, GetSIPAccount_External, SIPRequestAuthenticator.AuthenticateSIPRequest, FireProxyLogEvent, nonInviteTransaction);
nonInviteTransaction.NonInviteRequestReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(outgoingRequest); };
nonInviteTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else if (GetCanonicalDomain_External(sipRequest.URI.Host, true) != null)
{
// Call identified as incoming call for application server serviced domain.
if (sipRequest.URI.User.IsNullOrBlank())
{
// Cannot process incoming call if no user is specified.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, sipRequest.Method + " request received with an empty URI user " + sipRequest.URI.ToString() + ", returning address incomplete.", null));
SIPResponse addressIncompleteResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.AddressIncomplete, "No user specified");
m_sipTransport.SendResponse(addressIncompleteResponse);
}
else
{
// Send the incoming call to the call manager for processing.
string uriUser = sipRequest.URI.User;
string uriDomain = GetCanonicalDomain_External(sipRequest.URI.Host, true);
SIPNonInviteTransaction nonInviteTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, m_outboundProxy);
SIPNonInviteServerUserAgent incomingRequest = new SIPNonInviteServerUserAgent(m_sipTransport, m_outboundProxy, uriUser, uriDomain, SIPCallDirection.In, GetSIPAccount_External, null, FireProxyLogEvent, nonInviteTransaction);
nonInviteTransaction.NonInviteRequestReceived += (local, remote, transaction, request) => { m_callManager.QueueNewCall(incomingRequest); };
nonInviteTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
}
else
{
// Return not found for non-serviced domain.
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Domain not serviced " + sipRequest.URI.ToString() + ", returning not found.", null));
SIPResponse notServicedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotFound, "Domain not serviced");
m_sipTransport.SendResponse(notServicedResponse);
}
#endregion
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.UnrecognisedMessage, "MethodNotAllowed response for " + sipRequest.Method + " from " + fromUser + " socket " + remoteEndPoint.ToString() + ".", null));
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
catch (Exception excp)
{
string reqExcpError = "Exception SIPAppServerCore GotRequest (" + remoteEndPoint + "). " + excp.Message;
logger.Error(reqExcpError);
SIPMonitorEvent reqExcpEvent = new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.Error, reqExcpError, sipRequest, null, localSIPEndPoint, remoteEndPoint, SIPCallDirection.In);
FireProxyLogEvent(reqExcpEvent);
throw excp;
}
}
private void Agent_CallCancelled(ISIPServerUserAgent uas)
{
agent = null;
}
/// <summary>
/// Handler for processing incomign SIP requests.
/// </summary>
/// <param name="localSIPEndPoint">The end point the request was received on.</param>
/// <param name="remoteEndPoint">The end point the request came from.</param>
/// <param name="sipRequest">The SIP request received.</param>
private void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (m_uac != null && m_uac.SIPDialogue != null && sipRequest.Header.CallId == m_uac.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup by remote end.");
SIPNonInviteTransaction byeTransaction = m_sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
CallFinished();
}
else if (m_uas != null && m_uas.SIPDialogue != null && sipRequest.Header.CallId == m_uas.SIPDialogue.CallId)
{
// Call has been hungup by remote end.
StatusMessage("Call hungup.");
m_uas.SIPDialogue.Hangup(m_sipTransport, null);
CallFinished();
}
else
{
logger.Debug("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
StatusMessage("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
m_uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
m_uas.CallCancelled += UASCallCancelled;
IncomingCall();
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
StatusMessage("Call was cancelled by remote end.");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
logger.Debug("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
CallFinished();
}
else
{
logger.Debug("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
private static readonly int RTP_REPORTING_PERIOD_SECONDS = 5; // Period at which to write RTP stats.
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
IPAddress defaultAddr = LocalIPConfig.GetDefaultIPv4Address();
var sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new SIPTransactionEngine());
int port = FreePort.FindNextAvailableUDPPort(SIPConstants.DEFAULT_SIP_PORT);
var sipChannel = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));
sipTransport.AddSIPChannel(sipChannel);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource uasCts = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept a little it simpler.
uas?.Hangup();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uasCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(defaultAddr, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, uasCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup();
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup();
uasCts?.Cancel();
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
if (uas?.IsHungup == false)
{
uas?.Hangup();
}
uasCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
var rtpJob = (from job in m_rtpJobs.Values where job.UAS.CallRequest.Header.CallId == sipRequest.Header.CallId select job).FirstOrDefault();
if (rtpJob != null)
{
rtpJob.Stop();
// Call has been hungup by remote end.
Console.WriteLine("Call hungup by client: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
Publish(rtpJob.QueueName, "BYE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
//Console.WriteLine("Request Received " + localSIPEndPoint + "<-" + remoteEndPoint + "\n" + sipRequest.ToString());
//m_uas.SIPDialogue.Hangup(m_sipTransport, null);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
}
else
{
Console.WriteLine("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".\n");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Console.WriteLine("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
Publish(sipRequest.URI.User, "INVITE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
Console.WriteLine(sipRequest.Body);
SIPPacketMangler.MangleSIPRequest(SIPMonitorServerTypesEnum.Unknown, sipRequest, null, LogTraceMessage);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
var uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, LogTraceMessage, uasTransaction);
uas.CallCancelled += UASCallCancelled;
RTPDiagnosticsJob rtpJob = new RTPDiagnosticsJob(m_rtpListenIPAddress, m_publicIPAddress, uas, sipRequest);
string sdpAddress = SDP.GetSDPRTPEndPoint(sipRequest.Body).Address.ToString();
// Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.
IPEndPoint expectedRTPEndPoint = new IPEndPoint(rtpJob.RemoteRTPEndPoint.Address, rtpJob.RemoteRTPEndPoint.Port);
if (IPSocket.IsPrivateAddress(rtpJob.RemoteRTPEndPoint.Address.ToString()))
{
expectedRTPEndPoint.Address = remoteEndPoint.Address;
}
Publish(sipRequest.URI.User, "Advertised RTP remote socket " + rtpJob.RemoteRTPEndPoint + ", expecting from " + expectedRTPEndPoint + ".");
m_rtpJobs.Add(rtpJob.RTPListenEndPoint.Port, rtpJob);
//ThreadPool.QueueUserWorkItem(delegate { StartRTPListener(rtpJob); });
Console.WriteLine(rtpJob.LocalSDP.ToString());
uas.Answer("application/sdp", rtpJob.LocalSDP.ToString(), CallProperties.CreateNewTag(), null, SIPDialogueTransferModesEnum.NotAllowed);
var hangupTimer = new Timer(delegate
{
if (!rtpJob.StopJob)
{
if (uas != null && uas.SIPDialogue != null)
{
if (rtpJob.RTPPacketReceived && !rtpJob.ErrorOnRTPSend)
{
Publish(sipRequest.URI.User, "Test completed. There were no RTP send or receive errors.");
}
else if (!rtpJob.RTPPacketReceived)
{
Publish(sipRequest.URI.User, "Test completed. An error was identified, no RTP packets were received.");
}
else
{
Publish(sipRequest.URI.User, "Test completed. An error was identified, there was a problem when attempting to send an RTP packet.");
}
rtpJob.Stop();
uas.SIPDialogue.Hangup(m_sipTransport, null);
}
}
}, null, HANGUP_TIMEOUT, Timeout.Infinite);
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
Console.WriteLine("Matching CANCEL request received " + sipRequest.URI.ToString() + ".\n");
Publish(sipRequest.URI.User, "CANCEL request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
Console.WriteLine("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".\n");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else
{
Console.WriteLine("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.\n");
Publish(sipRequest.URI.User, sipRequest.Method + " request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
/// <summary>
/// Accepts an incoming call. This is the first step in answering a call.
/// From this point the call can still be rejected, redirected or answered.
/// </summary>
/// <param name="sipRequest">The SIP request containing the incoming call request.</param>
public void Accept(SIPRequest sipRequest)
{
m_pendingIncomingCall = m_userAgent.AcceptCall(sipRequest);
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2("localhost.pfx");
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.G722)))
{
Log.LogDebug($"Using G722 RTP media type and audio file {AUDIO_FILE_G722}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.G722, null, null);
audioFile = AUDIO_FILE_G722;
}
else if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(dstRtpEndPoint.AddressFamily == AddressFamily.InterNetworkV6 ? IPAddress.IPv6Any : IPAddress.Any, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
IPEndPoint rtpEndPoint = new IPEndPoint(rtpAddress, (rtpSocket.LocalEndPoint as IPEndPoint).Port);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, audioFile, rtpCts))
.ContinueWith(_ =>
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
/// <summary>
/// Cleans up after a SIP call has completely finished.
/// </summary>
private void CallFinished()
{
m_pendingIncomingCall = null;
CallEnded(this);
}
private void Transport_SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
var endpoint = new SIPEndPoint(SIPProtocolsEnum.udp, publicIPAddress, localSIPEndPoint.Port);
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
if (transaction != null)
{
return;
}
logger.DebugFormat("{0} Incoming call from {1}", prefix, sipRequest.Header.From.FromURI.User);
transaction = transport.CreateUASTransaction(sipRequest, remoteEndPoint, endpoint, null);
agent = new SIPServerUserAgent(
transport,
null,
sipRequest.Header.From.FromURI.User,
null,
SIPCallDirection.In,
null,
null,
null,
transaction);
agent.CallCancelled += Agent_CallCancelled;
agent.TransactionComplete += Agent_TransactionComplete;
agent.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
agent.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answer = SDP.ParseSDPDescription(agent.CallRequest.Body);
var address = IPAddress.Parse(answer.Connection.ConnectionAddress);
var port = answer.Media.FirstOrDefault(m => m.Media == SDPMediaTypesEnum.audio).Port;
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(address, port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
// TODO Fix hardcoded ports
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += Channel_OnFrameReady;
rtpChannel.Start();
// Send some setup parameters to punch a hole in the firewall/router
rtpChannel.SendRTPRaw(new byte[] { 80, 95, 198, 88, 55, 96, 225, 141, 215, 205, 185, 242, 00 });
rtpChannel.OnControlDataReceived += (b) => { logger.Debug($"{prefix} Control Data Received; {b.Length} bytes"); };
rtpChannel.OnControlSocketDisconnected += () => { logger.Debug($"{prefix} Control Socket Disconnected"); };
var announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
},
Port = rtpChannel.RTPPort
};
sdp.Media.Add(announcement);
SetState(State.Listening, sipRequest.Header.From.FromURI.User);
agent.Progress(SIPResponseStatusCodesEnum.Accepted, null, null, null, null);
agent.Answer(SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
SetState(State.Busy, "");
return;
}
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
if (State != State.Busy)
{
return;
}
logger.DebugFormat("{0} Hangup from {1}", prefix, sipRequest.Header.From.FromURI.User);
var noninvite = transport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, endpoint, null);
var response = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
noninvite.SendFinalResponse(response);
SetState(State.Finished, Endpoint);
rtpChannel.OnFrameReady -= Channel_OnFrameReady;
rtpChannel.Close();
agent.TransactionComplete -= Agent_TransactionComplete;
agent.CallCancelled -= Agent_CallCancelled;
agent = null;
transaction = null;
SetState(State.Ready, Endpoint);
return;
}
if (sipRequest.Method == SIPMethodsEnum.ACK)
{
}
if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
}
}
