/// <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.Microphone,
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);
}
}
private void ProbeWorkers()
{
try {
while (!m_exit)
{
try {
SIPEndPoint activeWorkerEndPoint = GetFirstHealthyEndPoint();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(m_dispatcherUsername, null, "sip:" + m_dispatcherUsername + "@" + activeWorkerEndPoint.SocketEndPoint.ToString(),
"sip:" + m_dispatcherUsername + "@sipcalldispatcher", "sip:" + activeWorkerEndPoint.SocketEndPoint.ToString(), null, null, null, SIPCallDirection.Out, null, null, null);
SIPClientUserAgent uac = new SIPClientUserAgent(m_sipTransport, null, null, null, null);
uac.CallAnswered += DispatcherCallAnswered;
uac.CallFailed += new SIPCallFailedDelegate(DispatcherCallFailed);
uac.Call(callDescriptor);
}
catch (Exception probeExcp) {
dispatcherLogger.Error("Exception SIPCallDispatcher Sending Probe. " + probeExcp.Message);
}
Thread.Sleep(PROBE_WORKER_CALL_PERIOD);
}
}
catch (Exception excp) {
logger.Error("Exception SIPCallDispatcher ProberWorkers. " + excp.Message);
}
}
/// <summary>
/// Starts a call based on a single multi forward call leg.
/// </summary>
/// <param name="calls">The list of simultaneous forwards to attempt.</param>
public void Start(List <SIPCallDescriptor> callDescriptors)
{
if (callDescriptors != null && callDescriptors.Count > 0)
{
for (int index = 0; index < callDescriptors.Count; index++)
{
int availableThreads = 0;
int ioCompletionThreadsAvailable = 0;
ThreadPool.GetAvailableThreads(out availableThreads, out ioCompletionThreadsAvailable);
if (availableThreads <= 0)
{
logger.Warn("The ThreadPool had no threads available in the pool to start a ForkCall leg, task will be queued.");
}
SIPCallDescriptor callDescriptor = callDescriptors[index];
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "ForkCall commencing call leg to " + callDescriptor.Uri + ".", m_username));
ThreadPool.QueueUserWorkItem(delegate { StartNewCallAsync(callDescriptor); });
}
}
else
{
CallLegCompleted();
}
}
static void Main(string[] args)
{
try
{
// Configure the SIP transport layer.
_sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new SIPTransactionEngine());
// Use default options to set up a SIP channel.
var localIP = LocalIPConfig.GetDefaultIPv4Address(); // Set this manually if needed.
int port = FreePort.FindNextAvailableUDPPort(_defaultSIPUdpPort);
var sipChannel = new SIPUDPChannel(new IPEndPoint(localIP, port));
_sipTransport.AddSIPChannel(sipChannel);
SIPCallDescriptor callDescriptor = new SIPCallDescriptor("test", null, "sip:[email protected]", "sip:[email protected]", null, null, null, null, SIPCallDirection.Out, null, null, null);
SIPNonInviteClientUserAgent notifyUac = new SIPNonInviteClientUserAgent(_sipTransport, null, callDescriptor, null, null, (monitorEvent) => { Console.WriteLine("Debug: " + monitorEvent.Message); });
notifyUac.ResponseReceived += (resp) => { Console.WriteLine(resp.ToString()); };
notifyUac.SendRequest(SIPMethodsEnum.NOTIFY);
ManualResetEvent mre = new ManualResetEvent(false);
mre.WaitOne();
}
catch (Exception excp)
{
Console.WriteLine("Exception Main. " + excp);
}
finally
{
Console.WriteLine("Press any key to exit...");
Console.ReadLine();
}
}
/// <summary>
/// Fired after each call leg forward attempt is completed.
/// </summary>
private void CallLegCompleted()
{
try
{
if (!m_callAnswered && !m_commandCancelled)
{
if (m_switchCalls.Count > 0 || m_delayedCalls.Count > 0)
{
// There are still calls on this leg in progress.
// If there are no current calls then start the next delayed one.
if (m_switchCalls.Count == 0)
{
SIPCallDescriptor nextCall = null;
lock (m_delayedCalls)
{
foreach (SIPCallDescriptor call in m_delayedCalls)
{
if (nextCall == null || nextCall.DelaySeconds > call.DelaySeconds)
{
nextCall = call;
}
}
}
if (nextCall != null)
{
nextCall.DelayMRE.Set();
}
}
}
else if (m_priorityCallsQueue.Count != 0 && !m_callAnswered)
{
List <SIPCallDescriptor> nextPrioritycalls = m_priorityCallsQueue.Dequeue();
Start(nextPrioritycalls);
}
else if (CallFailed != null)
{
// No more call legs to attempt, or call has already been answered or cancelled.
if (m_lastFailureStatus != SIPResponseStatusCodesEnum.None)
{
CallFailed(m_lastFailureStatus, m_lastFailureReason, null);
}
else
{
CallFailed(SIPResponseStatusCodesEnum.TemporarilyUnavailable, "All forwards failed.", null);
}
}
}
}
catch (Exception excp)
{
logger.Error("Exception CallLegCompleted. " + excp);
}
}
public void Start(string endpoint)
{
this.endpoint = endpoint;
var caller = "1003";
var password = passwords[0];
var port = FreePort.FindNextAvailableUDPPort(15090);
rtpChannel = new RTPChannel
{
DontTimeout = true,
RemoteEndPoint = new IPEndPoint(IPAddress.Parse(asterisk), port)
};
rtpChannel.SetFrameType(FrameTypesEnum.Audio);
rtpChannel.ReservePorts(15000, 15090);
rtpChannel.OnFrameReady += RtpChannel_OnFrameReady;
uac = new SIPClientUserAgent(transport, null, null, null, null);
var uri = SIPURI.ParseSIPURIRelaxed($"{ endpoint }@{ asterisk }");
var from = (new SIPFromHeader(caller, new SIPURI(caller, asterisk, null), null)).ToString();
var random = Crypto.GetRandomInt(5).ToString();
var sdp = new SDP
{
Version = 2,
Username = "******",
SessionId = random,
Address = localIPEndPoint.Address.ToString(),
SessionName = "redfox_" + random,
Timing = "0 0",
Connection = new SDPConnectionInformation(publicIPAddress.ToString())
};
var announcement = new SDPMediaAnnouncement
{
Media = SDPMediaTypesEnum.audio,
MediaFormats = new List <SDPMediaFormat>()
{
new SDPMediaFormat((int)SDPMediaFormatsEnum.PCMU, "PCMU", 8000)
},
Port = rtpChannel.RTPPort
};
sdp.Media.Add(announcement);
var descriptor = new SIPCallDescriptor(caller, password, uri.ToString(), from, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null);
uac.CallTrying += Uac_CallTrying;
uac.CallRinging += Uac_CallRinging;
uac.CallAnswered += Uac_CallAnswered;
uac.CallFailed += Uac_CallFailed;
uac.Call(descriptor);
}
static async Task DialNumber()
{
string fromHeader = (new SIPFromHeader(USERNAME, new SIPURI(USERNAME, DOMAIN, null), null)).ToString();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(USERNAME, PASSWORD, DEFAULT_CALL_DESTINATION, fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, null, null);
callDescriptor.CallId = "12028883999";
userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallRinging += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Failed: {err}, Status code: {resp?.StatusCode}");
userAgent.ClientCallAnswered += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.OnDtmfTone += (key, duration) => OnDtmfTone(userAgent, key, duration);
userAgent.OnRtpEvent += (evt, hdr) => Console.WriteLine($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
userAgent.OnCallHungup += OnHangup;
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var callResult = await userAgent.Call(callDescriptor, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
if (callResult)
{
Console.WriteLine("Enter digits one after another");
for (int i = 0; i < 11; i++)
{
var p = Console.ReadLine();
await userAgent.SendDtmf(byte.Parse(p));
}
}
Console.WriteLine("Enter ?");
Console.ReadLine();
await userAgent.SendDtmf(35);
Thread.Sleep(60000);
userAgent.Hangup();
_waveFile.Dispose();
Console.WriteLine("Hangup");
}
/// <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);
}
}
/// <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>
/// Gets the call descriptor to allow an outgoing call to be placed.
/// </summary>
/// <param name="callUri">The URI to place the call to.</param>
/// <param name="rtpSession">The RTP session that will be handling the RTP/RTCP packets for the call.</param>
/// <returns>A call descriptor.</returns>
private static SIPCallDescriptor GetCallDescriptor(string callUri)
{
// Create a call descriptor to place an outgoing call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri,
$"sip:{SIP_USERNAME}@localhost",
callUri,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
null,
null);
return(callDescriptor);
}
public void CallWithAdjustedInviteHeaderTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
MockSIPChannel channel = new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0));
transport.AddSIPChannel(channel);
SIPClientUserAgent userAgent = new SIPClientUserAgent(
transport,
new SIPEndPoint(new IPEndPoint(new IPAddress(new byte[] { 192, 168, 11, 50 }), 5060)),
"owner",
"admin",
null);
SIPContactHeader testHeader = new SIPContactHeader("Contact Name", new SIPURI("User", "Host", "Param=Value"));
userAgent.AdjustInvite = invite =>
{
invite.Header.Contact = new List <SIPContactHeader>
{
testHeader
};
return(invite);
};
var desc = new SIPCallDescriptor(
"user",
"pass",
"sip:user@host",
"sip:user@host",
"sip:destination@destinationhost",
null,
new List <string>(),
"user",
SIPCallDirection.Out,
null,
null,
null);
userAgent.Call(desc);
channel.SIPMessageSent.WaitOne(5000);
Assert.Contains(testHeader.ToString(), channel.LastSIPMessageSent);
}
private void StartNewCallAsync(SIPCallDescriptor callDescriptor)
{
try
{
callDescriptor.DialPlanContextID = (m_dialPlanContext != null) ? m_dialPlanContext.DialPlanContextID : Guid.Empty;
if (Thread.CurrentThread.Name == null)
{
Thread.CurrentThread.Name = THEAD_NAME + DateTime.Now.ToString("HHmmss") + "-" + Crypto.GetRandomString(3);
}
StartNewCallSync(callDescriptor);
}
catch (Exception excp)
{
logger.Error("Exception StartNewCallAsync. " + excp.Message);
}
}
/// <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 void Call(MediaManager mediaManager, string destination)
{
//_initialisationTask.Wait(_cancelCallTokenSource.Token);
_mediaManager = mediaManager;
_mediaManager.NewCall();
// 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("Starting call to " + callURI.ToString() + ".");
m_uac = new SIPClientUserAgent(m_sipTransport, null, null, null, null);
m_uac.CallTrying += CallTrying;
m_uac.CallRinging += CallRinging;
m_uac.CallAnswered += CallAnswered;
m_uac.CallFailed += CallFailed;
// Get the SDP requesting that the public IP address be used if the host on the call destination is not a private IP address.
SDP sdp = _mediaManager.GetSDP(!(IPAddress.TryParse(callURI.Host, out _) && IPSocket.IsPrivateAddress(callURI.Host)));
System.Diagnostics.Debug.WriteLine(sdp.ToString());
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, _sdpMimeContentType, sdp.ToString(), null);
m_uac.Call(callDescriptor);
}
public void Call(SIPCallDescriptor descriptor)
{
try
{
CallDescriptor = descriptor;
SIPURI destinationURI = SIPURI.ParseSIPURIRelaxed(descriptor.Uri);
bool wasSDPMangled = false;
IPEndPoint sdpEndPoint = null;
if (descriptor.MangleIPAddress != null)
{
sdpEndPoint = SDP.GetSDPRTPEndPoint(descriptor.Content);
if (sdpEndPoint != null)
{
descriptor.Content = SIPPacketMangler.MangleSDP(descriptor.Content, descriptor.MangleIPAddress.ToString(), out wasSDPMangled);
}
}
if (wasSDPMangled)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "SDP on Google Voice call had RTP socket mangled from " + sdpEndPoint.ToString() + " to " + descriptor.MangleIPAddress.ToString() + ":" + sdpEndPoint.Port + ".", Owner));
}
else if (sdpEndPoint != null)
{
Log_External(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "SDP on Google Voice call could not be mangled, using original RTP socket of " + sdpEndPoint.ToString() + ".", Owner));
}
SIPDialogue = m_googleVoiceCall.InitiateCall(descriptor.Username, descriptor.Password, descriptor.CallbackNumber, destinationURI.User, descriptor.CallbackPattern, descriptor.CallbackPhoneType, MAX_CALLBACK_WAIT_TIME, descriptor.ContentType, descriptor.Content);
if (SIPDialogue != null)
{
CallAnswered(this, null);
}
else
{
CallFailed(this, "Google Voice call failed.");
}
}
catch (Exception excp)
{
logger.Error("Exception GoogleVoiceCallAgent Call. " + excp.Message);
CallFailed(this, excp.Message);
}
}
/// <summary>
/// Sends the SIP INVITE probe request.
/// </summary>
private void ProbeWorker(SIPAppServerWorker worker, bool isInitialProbe)
{
try
{
if (isInitialProbe)
{
worker.InitialProbeCount++;
}
int workerProcessID = worker.WorkerProcess.Id;
SIPEndPoint workerEndPoint = worker.AppServerEndpoint;
DateTime probeSentAt = DateTime.Now;
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(m_dispatcherUsername, null, "sip:" + m_dispatcherUsername + "@" + workerEndPoint.GetIPEndPoint().ToString(),
"sip:" + m_dispatcherUsername + "@sipcalldispatcher", "sip:" + workerEndPoint.GetIPEndPoint().ToString(), null, null, null, SIPCallDirection.Out, null, null, null);
SIPClientUserAgent uac = new SIPClientUserAgent(m_sipTransport, null, null, null, null);
uac.CallFailed += (failedUAC, errorMessage) =>
{
AppServerCallFailed(failedUAC, errorMessage, workerProcessID, probeSentAt, isInitialProbe);
};
uac.CallAnswered += (call, sipResponse) =>
{
if (sipResponse.Status != SIPResponseStatusCodesEnum.BadExtension)
{
//logger.Warn("Probe call answered with unexpected response code of " + sipResponse.StatusCode + ".");
AppServerCallFailed(call, "Unexpected response of " + ((int)sipResponse.StatusCode) + " on probe call.", workerProcessID, probeSentAt, isInitialProbe);
}
else
{
AppServerCallSucceeded(call);
}
};
uac.Call(callDescriptor);
}
catch (Exception excp)
{
logger.Error("Exception SIPAppServerManager ProberWorker. " + excp.Message);
}
}
/// <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 void 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);
}
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("Starting call to " + callURI.ToString() + ".");
m_uac = new SIPClientUserAgent(m_sipTransport, null, null, null, null);
m_uac.CallTrying += CallTrying;
m_uac.CallRinging += CallRinging;
m_uac.CallAnswered += CallAnswered;
m_uac.CallFailed += CallFailed;
_audioChannel = new AudioChannel();
// Get the SDP requesting that the public IP address be used if the host on the call destination is not a private IP address.
SDP sdp = _audioChannel.GetSDP(!(IPSocket.IsIPAddress(callURI.Host) && IPSocket.IsPrivateAddress(callURI.Host)));
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(sipUsername, sipPassword, callURI.ToString(), fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, sdp.ToString(), null);
m_uac.Call(callDescriptor);
}
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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"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 RTPSession((int)SDPMediaFormatsEnum.PCMU, null, null, true, 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, RTPSession.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, RTPSession.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, RTPSession.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();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0], out callUri))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var audioSession = new WindowsAudioEndPoint(new AudioEncoder());
var rtpSession = new VoIPMediaSession(audioSession.ToMediaEndPoints());
var offerSDP = rtpSession.CreateOffer(null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += async(uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
if (resp.Status == SIPResponseStatusCodesEnum.SessionProgress)
{
await rtpSession.Start();
}
};
uac.CallFailed += (uac, err, resp) =>
{
Log.LogWarning($"Call attempt to {uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += async(iuac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body));
if (result == SetDescriptionResultEnum.OK)
{
await rtpSession.Start();
}
else
{
Log.LogWarning($"Failed to set remote description {result}.");
uac.Hangup();
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(okResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
exitMre.Set();
}
}
};
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
offerSDP.ToString(),
null);
uac.Call(callDescriptor, null);
uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
rtpSession.Close(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
/// <summary>
/// Establishes a new call with the client end tied to the proxy. Since the proxy will not be sending any audio the idea is that once
/// the call is up it should be re-INVITED off somewhere else pronto to avoid the callee sitting their listening to dead air.
/// </summary>
/// <param name="dest1">The dial string of the first call to place.</param>
/// <param name="dest2">The dial string of the second call to place.</param>
/// <param name="delaySeconds">Delay in seconds before placing the first call. Gives the user a chance to hangup their phone if they are calling themselves back.</param>
/// <param name="ringTimeoutLeg1">The ring timeout for the first call leg, If 0 the max timeout will be used.</param>
/// <param name="ringTimeoutLeg1">The ring timeout for the second call leg, If 0 the max timeout will be used.</param>
/// <param name="customHeadersCallLeg1">A | delimited string that contains a list of custom SIP headers to add to the INVITE request sent for the first call leg.</param>
/// /// <param name="customHeadersCallLeg2">A | delimited string that contains a list of custom SIP headers to add to the INVITE request sent for the second call leg.</param>
/// <returns>The result of the call.</returns>
public void Callback(string dest1, string dest2, int delaySeconds, int ringTimeoutLeg1, int ringTimeoutLeg2, string customHeadersCallLeg1, string customHeadersCallLeg2)
{
var ts = new CancellationTokenSource();
CancellationToken ct = ts.Token;
try
{
if (delaySeconds > 0)
{
delaySeconds = (delaySeconds > MAXCALLBACK_DELAY_SECONDS) ? MAXCALLBACK_DELAY_SECONDS : delaySeconds;
Log("Callback app delaying by " + delaySeconds + "s.");
Thread.Sleep(delaySeconds * 1000);
}
Log("Callback app commencing first leg to " + dest1 + ".");
SIPEndPoint defaultUDPEP = m_sipTransport.GetDefaultSIPEndPoint(SIPProtocolsEnum.udp);
SIPRequest firstLegDummyInviteRequest = GetCallbackInviteRequest(defaultUDPEP.GetIPEndPoint(), null);
ForkCall firstLegCall = new ForkCall(m_sipTransport, Log_External, m_callManager.QueueNewCall, null, m_username, m_adminMemberId, m_outboundProxy, m_callManager, null);
m_firstLegDialogue = Dial(firstLegCall, dest1, ringTimeoutLeg1, 0, firstLegDummyInviteRequest, SIPCallDescriptor.ParseCustomHeaders(customHeadersCallLeg1));
if (m_firstLegDialogue == null)
{
Log("The first call leg to " + dest1 + " was unsuccessful.");
return;
}
// Persist the dialogue to the database so any hangup can be detected.
m_sipDialoguePersistor.Add(new SIPDialogueAsset(m_firstLegDialogue));
SDP firstLegSDP = SDP.ParseSDPDescription(m_firstLegDialogue.RemoteSDP);
string call1SDPIPAddress = firstLegSDP.Connection.ConnectionAddress;
int call1SDPPort = firstLegSDP.Media[0].Port;
Log("The first call leg to " + dest1 + " was successful, audio socket=" + call1SDPIPAddress + ":" + call1SDPPort + ".");
Log("Callback app commencing second leg to " + dest2 + ".");
SIPRequest secondLegDummyInviteRequest = GetCallbackInviteRequest(defaultUDPEP.GetIPEndPoint(), m_firstLegDialogue.RemoteSDP);
ForkCall secondLegCall = new ForkCall(m_sipTransport, Log_External, m_callManager.QueueNewCall, null, m_username, m_adminMemberId, m_outboundProxy, m_callManager, null);
Task.Factory.StartNew(() =>
{
while (true)
{
Thread.Sleep(CHECK_FIRST_LEG_FOR_HANGUP_PERIOD);
Console.WriteLine("Checking if first call leg is still up...");
if (ct.IsCancellationRequested)
{
Console.WriteLine("Checking first call leg task was cancelled.");
break;
}
else
{
// Check that the first call leg hasn't been hung up.
var dialog = m_sipDialoguePersistor.Get(m_firstLegDialogue.Id);
if (dialog == null)
{
Console.WriteLine("First call leg has been hungup.");
// The first call leg has been hungup while waiting for the second call.
Log("The first call leg was hungup while the second call leg was waiting for an answer.");
secondLegCall.CancelNotRequiredCallLegs(CallCancelCause.ClientCancelled);
break;
}
}
}
Console.WriteLine("Checking first call leg task finished...");
}, ct);
SIPDialogue secondLegDialogue = Dial(secondLegCall, dest2, ringTimeoutLeg2, 0, secondLegDummyInviteRequest, SIPCallDescriptor.ParseCustomHeaders(customHeadersCallLeg2));
ts.Cancel();
if (secondLegDialogue == null)
{
Log("The second call leg to " + dest2 + " was unsuccessful.");
m_firstLegDialogue.Hangup(m_sipTransport, m_outboundProxy);
return;
}
// Check that the first call leg hasn't been hung up.
var firstLegDialog = m_sipDialoguePersistor.Get(m_firstLegDialogue.Id);
if (firstLegDialog == null)
{
// The first call leg has been hungup while waiting for the second call.
Log("The first call leg was hungup while waiting for the second call leg.");
secondLegDialogue.Hangup(m_sipTransport, m_outboundProxy);
return;
}
SDP secondLegSDP = SDP.ParseSDPDescription(secondLegDialogue.RemoteSDP);
string call2SDPIPAddress = secondLegSDP.Connection.ConnectionAddress;
int call2SDPPort = secondLegSDP.Media[0].Port;
Log("The second call leg to " + dest2 + " was successful, audio socket=" + call2SDPIPAddress + ":" + call2SDPPort + ".");
// Persist the second leg dialogue and update the bridge ID on the first call leg.
Guid bridgeId = Guid.NewGuid();
secondLegDialogue.BridgeId = bridgeId;
m_sipDialoguePersistor.Add(new SIPDialogueAsset(secondLegDialogue));
m_sipDialoguePersistor.UpdateProperty(firstLegDialog.Id, "BridgeID", bridgeId.ToString());
//m_callManager.CreateDialogueBridge(m_firstLegDialogue, secondLegDialogue, m_username);
Log("Re-inviting Callback dialogues to each other.");
m_callManager.ReInvite(m_firstLegDialogue, secondLegDialogue);
//m_callManager.ReInvite(secondLegDialogue, m_firstLegDialogue.RemoteSDP);
SendRTPPacket(call2SDPIPAddress + ":" + call2SDPPort, call1SDPIPAddress + ":" + call1SDPPort);
SendRTPPacket(call1SDPIPAddress + ":" + call1SDPPort, call2SDPIPAddress + ":" + call2SDPPort);
}
catch (Exception excp)
{
logger.Error("Exception CallbackApp. " + excp);
Log("Exception in Callback. " + excp);
}
finally
{
if (!ts.IsCancellationRequested)
{
ts.Cancel();
}
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
// Check whether an override desination has been entered on the command line.
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($"Call destination {callUri}.");
// Set up a default SIP transport.
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
EnableTraceLogs(_sipTransport);
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"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.
_ourRtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out _ourRtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
// Create a client/server user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
userAgent.ClientCallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
exitCts.Cancel();
};
userAgent.ClientCallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
// 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}.");
}
};
userAgent.CallHungup += () =>
{
Log.LogInformation($"Call hungup by remote party.");
exitCts.Cancel();
};
userAgent.OnReinviteRequest += ReinviteRequestReceived;
// The only incoming requests that need to be explicitly in this example program are in-dialog
// re-INVITE requests that are being used to place the call on/off hold.
_sipTransport.SIPTransportRequestReceived += (localSIPEndPoint, remoteEndPoint, sipRequest) =>
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
_sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception excp)
{
Log.LogError($"Exception processing request. {excp.Message}");
}
};
// 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(_ourRtpSocket, rtpRecvSession, exitCts));
Task.Run(() => SendRtp(_ourRtpSocket, rtpSendSession, exitCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri.ToString(),
$"sip:{SIP_USERNAME}@localhost",
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
_ourSDP.ToString(),
null);
userAgent.Call(callDescriptor);
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h')
{
// Place call on/off hold.
if (userAgent.IsAnswered)
{
if (_holdStatus == HoldStatus.None)
{
Log.LogInformation("Placing the remote call party on hold.");
_holdStatus = HoldStatus.WePutOnHold;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDONLY);
userAgent.SendReInviteRequest(_ourSDP);
}
else if (_holdStatus == HoldStatus.WePutOnHold)
{
Log.LogInformation("Removing the remote call party from hold.");
_holdStatus = HoldStatus.None;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
userAgent.SendReInviteRequest(_ourSDP);
}
else
{
Log.LogInformation("Sorry we're already on hold by the remote call party.");
}
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.Token.WaitHandle.WaitOne();
#region Cleanup.
Log.LogInformation("Exiting...");
_ourRtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && userAgent != null)
{
if (userAgent.IsAnswered)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {userAgent?.CallDescriptor?.To}.");
userAgent.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (_sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
_sipTransport.Shutdown();
}
#endregion
}
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($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
// Un/comment this line to see/hide each SIP message sent and received.
EnableTraceLogs(sipTransport);
// Note this relies on the callURI host being an IP address. If it's a hostname a DNS lookup is required.
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(callUri.ToSIPEndPoint().Address);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{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, 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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"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}.");
Log.LogDebug(resp.ToString());
};
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 is established. We'll wait for a few seconds and then transfer.
Task.Delay(DELAY_UNTIL_TRANSFER_MILLISECONDS).Wait();
SIPRequest referRequest = GetReferRequest(uac, SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI));
SIPNonInviteTransaction referTx = sipTransport.CreateNonInviteTransaction(referRequest, referRequest.RemoteSIPEndPoint, referRequest.LocalSIPEndPoint, null);
referTx.NonInviteTransactionFinalResponseReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPTransaction sipTransaction, SIPResponse sipResponse) =>
{
if (sipResponse.Header.CSeqMethod == SIPMethodsEnum.REFER && sipResponse.Status == SIPResponseStatusCodesEnum.Accepted)
{
Log.LogInformation("Call transfer was accepted by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
};
referTx.SendReliableRequest();
// At this point the call transfer 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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main()
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP trnasport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, 0)));
EnableTraceLogs(sipTransport);
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{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;
exitCts.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, exitCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, exitCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIP_USERNAME,
SIP_PASSWORD,
callUri.ToString(),
$"sip:{SIP_USERNAME}@localhost",
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h')
{
}
else if (keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
uac.Hangup();
exitCts.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitCts.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();
}
}
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($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var rtpSession = new RtpAVSession(dstAddress.AddressFamily, new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, null);
var offerSDP = await rtpSession.createOffer(new RTCOfferOptions { RemoteSignallingAddress = dstAddress });
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{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();
}
}
private void StartNewCallSync(SIPCallDescriptor callDescriptor)
{
try
{
callDescriptor.DialPlanContextID = (m_dialPlanContext != null) ? m_dialPlanContext.DialPlanContextID : Guid.Empty;
if (callDescriptor.DelaySeconds != 0)
{
callDescriptor.DelayMRE = new ManualResetEvent(false);
lock (m_delayedCalls)
{
m_delayedCalls.Add(callDescriptor);
}
int delaySeconds = (callDescriptor.DelaySeconds > MAX_DELAY_SECONDS) ? MAX_DELAY_SECONDS : callDescriptor.DelaySeconds;
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Delaying call leg to " + callDescriptor.Uri + " by " + delaySeconds + "s.", m_username));
callDescriptor.DelayMRE.WaitOne(delaySeconds * 1000);
}
lock (m_delayedCalls)
{
m_delayedCalls.Remove(callDescriptor);
}
if (!m_callAnswered && !m_commandCancelled)
{
ISIPClientUserAgent uacCall = null;
if (callDescriptor.ToSIPAccount == null)
{
if (callDescriptor.IsGoogleVoiceCall)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Creating Google Voice user agent for " + callDescriptor.Uri + ".", m_username));
uacCall = new GoogleVoiceUserAgent(m_sipTransport, m_callManager, m_statefulProxyLogEvent, m_username, m_adminMemberId, m_outboundProxySocket);
}
else
{
uacCall = new SIPClientUserAgent(m_sipTransport, m_outboundProxySocket, m_username, m_adminMemberId, m_statefulProxyLogEvent,
m_customerAccountDataLayer.GetRtccCustomer, m_customerAccountDataLayer.GetRtccRate, m_customerAccountDataLayer.GetBalance,
m_customerAccountDataLayer.ReserveInitialCredit, m_customerAccountDataLayer.UpdateRealTimeCallControlCDRID);
}
}
else
{
if (QueueNewCall_External == null)
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "B2B calls are not supported in this dialplan manifestation.", m_username));
}
else
{
FireProxyLogEvent(new SIPMonitorConsoleEvent(SIPMonitorServerTypesEnum.AppServer, SIPMonitorEventTypesEnum.DialPlan, "Creating B2B call for " + callDescriptor.Uri + ".", m_username));
uacCall = new SIPB2BUserAgent(m_statefulProxyLogEvent, QueueNewCall_External, m_sipTransport, m_username, m_adminMemberId);
}
}
//ISIPClientUserAgent uacCall = new JingleUserAgent(m_username, m_adminMemberId, m_statefulProxyLogEvent);
if (uacCall != null)
{
lock (m_switchCalls)
{
m_switchCalls.Add(uacCall);
}
uacCall.CallAnswered += UACCallAnswered;
uacCall.CallFailed += UACCallFailed;
uacCall.CallRinging += UACCallProgress;
//uacCall.CallTrying += UACCallTrying;
uacCall.Call(callDescriptor);
}
}
}
catch (Exception excp)
{
logger.Error("Exception ForkCall StartNewCall. " + excp.Message);
}
}
/// <summary>
/// Sends the SIP INVITE probe request.
/// </summary>
public void SendProbe()
{
try
{
if (WorkerProcess == null)
{
logger.Debug("When attempting to send probe the worker process was null. Marking for immediate restart.");
NeedsImmediateRestart = true;
}
else if (WorkerProcess.HasExited)
{
logger.Debug("When attempting to send probe the worker had exited. Marking for immediate restart.");
NeedsImmediateRestart = true;
}
else if (m_probeUAC != null && !m_probeUAC.IsUACAnswered)
{
// A probe call has timed out.
m_probeUAC.Cancel();
m_missedProbes++;
if (m_missedProbes >= m_missedProbesLimit)
{
logger.Warn(m_missedProbes + " probes missed for " + AppServerEndpoint.ToString() + ". Marking for immediate restart.");
NeedsImmediateRestart = true;
}
}
if (!NeedsImmediateRestart && !NeedsToRestart)
{
m_probeCount++;
//logger.Debug("Sending probe " + m_probeCount + " to " + AppServerEndpoint.GetIPEndPoint().ToString() + ".");
DateTime probeSentAt = DateTime.Now;
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(m_dispatcherUsername, null, "sip:" + m_dispatcherUsername + "@" + AppServerEndpoint.GetIPEndPoint().ToString(),
"sip:" + m_dispatcherUsername + "@sipcalldispatcher", "sip:" + AppServerEndpoint.GetIPEndPoint().ToString(), null, null, null, SIPCallDirection.Out, null, null, null);
m_probeUAC = new SIPClientUserAgent(m_sipTransport, null, null, null, null, null, null, null, null, null);
m_probeUAC.CallAnswered += (call, sipResponse) =>
{
//logger.Debug("Probe response received for " + AppServerEndpoint.ToString() + ".");
if (sipResponse.Status != SIPResponseStatusCodesEnum.BadExtension)
//if (sipResponse.Status != SIPResponseStatusCodesEnum.InternalServerError)
{
logger.Warn("Probe to " + AppServerEndpoint.ToString() + " answered incorrectly on probe number " + m_probeCount + " after "
+ DateTime.Now.Subtract(probeSentAt).TotalSeconds.ToString("0.##") + "s, unexpected response of " + ((int)sipResponse.StatusCode) + ".");
NeedsImmediateRestart = true;
}
else
{
m_gotInitialProbeResponse = true;
}
if (m_initialResponseMRE != null)
{
m_initialResponseMRE.Set();
}
};
m_probeUAC.Call(callDescriptor);
}
}
catch (Exception excp)
{
logger.Error("Exception SendProbe. " + 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($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
IPAddress localAddress = sipTransport.GetLocalAddress(IPAddress.Parse("8.8.8.8"));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(localAddress, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, port)));
//EnableTraceLogs(sipTransport);
// Select the IP address to use for RTP based on the destination SIP URI.
var endPointForCall = callUri.ToSIPEndPoint() == null?sipTransport.GetDefaultSIPEndPoint(callUri.Protocol) : sipTransport.GetDefaultSIPEndPoint(callUri.ToSIPEndPoint());
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
// TODO (find something better): If the SIP endpoint is using 0.0.0.0 for SIP use loopback for RTP.
IPAddress rtpAddress = localAddress;
NetServices.CreateRtpSocket(rtpAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{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($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
IPEndPoint remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Sending initial RTP packet to remote RTP socket {remoteRtpEndPoint}.");
// Send a dummy packet to open the NAT session on the RTP path.
rtpSendSession.SendAudioFrame(rtpSocket, remoteRtpEndPoint, 0, new byte[] { 0x00 });
}
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(() => SendRecvRtp(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();
}
}
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 example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource cts = new CancellationTokenSource();
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;
// 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 + 2);
var sipChannel = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));
sipTransport.AddSIPChannel(sipChannel);
// 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);
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) => SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += (uac, resp) => SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
SIPSorcery.Sys.Log.Logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
IPEndPoint remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
SIPSorcery.Sys.Log.Logger.LogDebug($"Sending initial RTP packet to remote RTP socket {remoteRtpEndPoint}.");
// Send a dummy packet to open the NAT session on the RTP path.
rtpSendSession.SendAudioFrame(rtpSocket, remoteRtpEndPoint, 0, new byte[] { 0x00 });
}
else
{
SIPSorcery.Sys.Log.Logger.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)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
cts.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(() => SendRecvRtp(rtpSocket, rtpSendSession, cts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
DESTINATION_SIP_URI,
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// 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.
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += async delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
cts.Cancel();
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to clean up...");
await Task.Delay(1000);
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
CancellationTokenSource exitCts = new CancellationTokenSource();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, OUTBOUND_PROXY);
userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
userAgent.OnCallHungup += (dialog) => exitCts.Cancel();
var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;
string fromHeader = (new SIPFromHeader(USERNAME, new SIPURI(USERNAME, DOMAIN, null), null)).ToString();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(USERNAME, PASSWORD, DESTINATION, fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, null, null);
//callDescriptor.CallId = "16152412565";
//callDescriptor.AuthUsername = USERNAME;
// Place the call and wait for the result.
var callTask = userAgent.Call(callDescriptor, voipMediaSession);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent != null)
{
if (userAgent.IsCalling || userAgent.IsRinging)
{
Console.WriteLine("Cancelling in progress call.");
userAgent.Cancel();
}
else if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up established call.");
userAgent.Hangup();
_waveFile.Dispose();
}
}
;
exitCts.Cancel();
};
Console.WriteLine("press ctrl-c to exit...");
bool callResult = await callTask;
if (callResult)
{
Console.WriteLine("Enter digits one after another");
string meetingNo = "1711622132";
Console.WriteLine("Enter meetingno ?");
Console.ReadLine();
foreach (var item in meetingNo)
{
await userAgent.SendDtmf(byte.Parse(item.ToString()));
Console.WriteLine("Sending DTMF - " + byte.Parse(item.ToString()));
Thread.Sleep(2000);
}
Thread.Sleep(2000);
await userAgent.SendDtmf(Encoding.ASCII.GetBytes("#")[0]);
Thread.Sleep(13000);
Console.WriteLine("Sending AttendeeID ?");
/*string attendeeId = "635619";
* foreach (var item in attendeeId)
* {
* await userAgent.SendDtmf(byte.Parse(item.ToString()));
* Console.WriteLine("Sending DTMF - " + byte.Parse(item.ToString()));
* Thread.Sleep(2000);
* }*/
await userAgent.SendDtmf(Encoding.ASCII.GetBytes("#")[0]);
Console.ReadLine();
await userAgent.SendDtmf(Encoding.ASCII.GetBytes("#")[0]);
Console.WriteLine($"Call to {DESTINATION} succeeded.");
exitCts.Token.WaitHandle.WaitOne();
}
else
{
Console.WriteLine($"Call to {DESTINATION} failed.");
}
Console.WriteLine("Exiting...");
if (userAgent?.IsHangingUp == true)
{
Console.WriteLine("Waiting 1s for the call hangup or cancel to complete...");
await Task.Delay(1000);
}
// Clean up.
sipTransport.Shutdown();
}
