private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Console.WriteLine("INVITE received from " + localSIPEndPoint.ToString());
IPEndPoint sipPhoneEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPServerUserAgent uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
SIPResponse tryingResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Trying, null);
uasTransaction.SendInformationalResponse(tryingResponse);
if (m_xmppClient == null)
{
m_xmppClient = new XMPPClient(XMPP_SERVER, XMPP_SERVER_PORT, XMPP_REALM, m_xmppUsername, m_xmppPassword);
m_xmppClient.Disconnected += XMPPDisconnected;
m_xmppClient.IsBound += () => { XMPPPlaceCall(uas); };
ThreadPool.QueueUserWorkItem(delegate { m_xmppClient.Connect(); });
}
else
{
XMPPPlaceCall(uas);
}
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
Console.WriteLine("Matching CANCEL request received " + sipRequest.URI.ToString() + ".");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
Console.WriteLine("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Console.WriteLine("BYE request received.");
if (m_activeCalls.ContainsKey(sipRequest.Header.CallId))
{
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
m_activeCalls[sipRequest.Header.CallId].TerminateXMPPCall();
m_activeCalls.Remove(sipRequest.Header.CallId);
}
else
{
SIPResponse doesntExistResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(doesntExistResponse);
}
}
}
/// <summary>
/// Sets relevant properties for this session based on the SDP from the remote party.
/// </summary>
/// <param name="sdp">The SDP from the remote call party.</param>
public void SetRemoteSDP(SDP sdp)
{
RemoteSDP = sdp;
foreach (var announcement in sdp.Media.Where(x => x.Media == SDPMediaTypesEnum.audio))
{
foreach (var mediaFormat in announcement.MediaFormats)
{
if (mediaFormat.FormatAttribute?.StartsWith(TELEPHONE_EVENT_ATTRIBUTE) == true)
{
if (!int.TryParse(mediaFormat.FormatID, out var remoteRtpEventPayloadID))
{
logger.LogWarning("The media format on the telephone event attribute was not a valid integer.");
}
else
{
base.RemoteRtpEventPayloadID = remoteRtpEventPayloadID;
}
break;
}
}
}
var dstEndPoint = sdp.GetSDPRTPEndPoint();
SetDestination(dstEndPoint, new IPEndPoint(dstEndPoint.Address, dstEndPoint.Port + 1));
}
public static string MangleSDP(string sdpBody, string publicIPAddress, out bool wasMangled)
{
wasMangled = false;
try
{
if (sdpBody != null && publicIPAddress != null)
{
string sdpAddress = SDP.GetSDPRTPEndPoint(sdpBody).Address.ToString();
// Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.
if (IPSocket.IsPrivateAddress(sdpAddress) && publicIPAddress != sdpAddress)
{
//logger.LogDebug("MangleSDP replacing private " + sdpAddress + " with " + publicIPAddress + ".");
string mangledSDP = Regex.Replace(sdpBody, @"c=IN IP4 (?<ipaddress>(\d+\.){3}\d+)", "c=IN IP4 " + publicIPAddress, RegexOptions.Singleline);
wasMangled = true;
return(mangledSDP);
}
}
else
{
logger.LogWarning("Mangle SDP was called with an empty body or public IP address.");
}
return(sdpBody);
}
catch (Exception excp)
{
logger.LogError("Exception MangleSDP. " + excp.Message);
return(sdpBody);
}
}
private void Answered(SDP xmppSDP)
{
//Console.WriteLine("Yay call answered.");
//Console.WriteLine(sdp.ToString());
m_xmppServerEndPoint = SDP.GetSDPRTPEndPoint(xmppSDP.ToString());
logger.Debug("Sending STUN binding request to " + m_xmppServerEndPoint + ".");
STUNMessage initMessage = new STUNMessage(STUNMessageTypesEnum.BindingRequest);
initMessage.AddUsernameAttribute(xmppSDP.IceUfrag + m_localSTUNUFrag);
byte[] stunMessageBytes = initMessage.ToByteBuffer();
m_xmppMediaSocket.Send(stunMessageBytes, stunMessageBytes.Length, m_xmppServerEndPoint);
m_uas.Answer("application/sdp", GetSDPForSIPResponse().ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
// private static ILogger logger = Log.Logger;
public static string MangleSDP(string sdpBody, string publicIPAddress, out bool wasMangled)
{
wasMangled = false;
try
{
if (sdpBody != null && publicIPAddress != null)
{
IPAddress addr = SDP.GetSDPRTPEndPoint(sdpBody).Address;
//rj2: need to consider publicAddress and IPv6 for mangling
IPAddress pubaddr = IPAddress.Parse(publicIPAddress);
string sdpAddress = addr.ToString();
// Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.
if (IPSocket.IsPrivateAddress(sdpAddress) && publicIPAddress != sdpAddress &&
pubaddr.AddressFamily == AddressFamily.InterNetworkV6 &&
addr.AddressFamily == AddressFamily.InterNetworkV6)
{
string mangledSDP = Regex.Replace(sdpBody, @"c=IN IP6 (?<ipaddress>([:a-fA-F0-9]+))",
"c=IN IP6" + publicIPAddress, RegexOptions.Singleline);
wasMangled = true;
return(mangledSDP);
}
else if (IPSocket.IsPrivateAddress(sdpAddress) && publicIPAddress != sdpAddress &&
pubaddr.AddressFamily ==
AddressFamily.InterNetwork &&
addr.AddressFamily == AddressFamily.InterNetwork)
{
//logger.LogDebug("MangleSDP replacing private " + sdpAddress + " with " + publicIPAddress + ".");
string mangledSDP = Regex.Replace(sdpBody, @"c=IN IP4 (?<ipaddress>(\d+\.){3}\d+)",
"c=IN IP4 " + publicIPAddress, RegexOptions.Singleline);
wasMangled = true;
return(mangledSDP);
}
}
else
{
Logger.Logger.Warn("Mangle SDP was called with an empty body or public IP address.");
}
return(sdpBody);
}
catch (Exception excp)
{
Logger.Logger.Error("Exception MangleSDP. ->" + excp.Message);
return(sdpBody);
}
}
/// <summary>
/// An outgoing call was successfully answered.
/// </summary>
/// <param name="uac">The local SIP user agent client that initiated the call.</param>
/// <param name="sipResponse">The SIP answer response received from the remote party.</param>
private void CallAnswered(ISIPClientUserAgent uac, SIPResponse sipResponse)
{
StatusMessage("Call answered: " + sipResponse.StatusCode + " " + sipResponse.ReasonPhrase + ".");
if (sipResponse.StatusCode >= 200 && sipResponse.StatusCode <= 299)
{
IPEndPoint remoteSDPEndPoint = SDP.GetSDPRTPEndPoint(sipResponse.Body);
_audioChannel.SetRemoteRTPEndPoint(remoteSDPEndPoint);
CallAnswer();
}
else
{
CallFinished();
}
}
/// <summary>
/// Event handler for an answer on an outgoing Google Voice call.
/// </summary>
/// <param name="xmppSDP">The SDP packet received from the Google Voice gateway.</param>
private void XMPPAnswered(SDP xmppSDP)
{
StatusMessage("Google Voice call answered.");
IPEndPoint remoteSDPEndPoint = SDP.GetSDPRTPEndPoint(xmppSDP.ToString());
_audioChannel.SetRemoteRTPEndPoint(remoteSDPEndPoint);
// Google Voice require that a STUN exchange occurs on the RTP socket before the RTP packet can flow.
// This code block sends a STUN binding request to the Google Voice gateway.
STUNMessage initMessage = new STUNMessage(STUNMessageTypesEnum.BindingRequest);
initMessage.AddUsernameAttribute(xmppSDP.IceUfrag + m_localSTUNUFrag);
byte[] stunMessageBytes = initMessage.ToByteBuffer();
//_audioChannel.SendRTPRaw(stunMessageBytes, stunMessageBytes.Length);
}
public void Call(SIPCallDescriptor sipCallDescriptor)
{
CallDescriptor = sipCallDescriptor;
m_xmppClient = new XMPPClient("talk.google.com", 5222, "google.com", null, null);
m_xmppClient.IsBound += IsBound;
m_xmppClient.Answered += Answered;
IPEndPoint sdpEndPoint = SDP.GetSDPRTPEndPoint(CallDescriptor.Content);
if (IPSocket.IsPrivateAddress(sdpEndPoint.Address.ToString()))
{
bool wasSDPMangled;
CallDescriptor.Content = SIPPacketMangler.MangleSDP(CallDescriptor.Content, CallDescriptor.MangleIPAddress.ToString(), out wasSDPMangled);
}
ThreadPool.QueueUserWorkItem(delegate { m_xmppClient.Connect(); });
}
/// <summary>
/// Event handler for receiving a re-INVITE request on an established call.
/// In call requests can be used for multitude of different purposes. In this
/// example program we're only concerned with re-INVITE requests being used
/// to place a call on/off hold.
/// </summary>
/// <param name="uasTransaction">The user agent server invite transaction that
/// was created for the request. It needs to be used for sending responses
/// to ensure reliable delivery.</param>
private static void ReinviteRequestReceived(UASInviteTransaction uasTransaction)
{
SIPRequest reinviteRequest = uasTransaction.TransactionRequest;
// Re-INVITEs can also be changing the RTP end point. We can update this each time.
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(reinviteRequest.Body);
_remoteRtpEndPoint = dstRtpEndPoint;
// If the RTP callfow attribute has changed it's most likely due to being placed on/off hold.
SDP newSDP = SDP.ParseSDPDescription(reinviteRequest.Body);
if (GetRTPStatusAttribute(newSDP) == RTP_ATTRIBUTE_SENDONLY)
{
Log.LogInformation("Remote call party has placed us on hold.");
_holdStatus = HoldStatus.RemotePutOnHold;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_RECVONLY);
var okResponse = SIPTransport.GetResponse(reinviteRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.ContentType = SDP.SDP_MIME_CONTENTTYPE;
okResponse.Body = _ourSDP.ToString();
uasTransaction.SendFinalResponse(okResponse);
}
else if (GetRTPStatusAttribute(newSDP) == RTP_ATTRIBUTE_SENDRECV && _holdStatus != HoldStatus.None)
{
Log.LogInformation("Remote call party has taken us off hold.");
_holdStatus = HoldStatus.None;
_ourSDP = GetSDP(_ourRtpSocket.LocalEndPoint as IPEndPoint, RTP_ATTRIBUTE_SENDRECV);
var okResponse = SIPTransport.GetResponse(reinviteRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.ContentType = SDP.SDP_MIME_CONTENTTYPE;
okResponse.Body = _ourSDP.ToString();
uasTransaction.SendFinalResponse(okResponse);
}
else
{
Log.LogWarning("Not sure what the remote call party wants us to do...");
// We'll just reply Ok and hope eveything is good.
var okResponse = SIPTransport.GetResponse(reinviteRequest, SIPResponseStatusCodesEnum.Ok, null);
okResponse.Header.ContentType = SDP.SDP_MIME_CONTENTTYPE;
okResponse.Body = _ourSDP.ToString();
uasTransaction.SendFinalResponse(okResponse);
}
}
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);
}
}
public void GetFirstMedaiOfferIPv6RTPSocketUnitTest()
{
Logger.LogDebug(System.Reflection.MethodBase.GetCurrentMethod().Name);
string sdpStr = "v=0" + m_CRLF +
"o=nasa1 971731711378798081 0 IN IP6 2201:056D::112E:144A:1E24" + m_CRLF +
"s=(Almost) live video feed from Mars-II satellite" + m_CRLF +
"p=+1 713 555 1234" + m_CRLF +
"c=IN IP6 FF1E:03AD::7F2E:172A:1E24" + m_CRLF +
"t=3338481189 3370017201" + m_CRLF +
"m=audio 6000 RTP/AVP 2" + m_CRLF +
"a=rtpmap:2 G726-32/8000" + m_CRLF +
"m=video 6024 RTP/AVP 107" + m_CRLF +
"a=rtpmap:107 H263-1998/90000";
IPEndPoint audioRtpEndPoint = SDP.GetSDPRTPEndPoint(sdpStr);
Assert.True(audioRtpEndPoint.Address.Equals(IPAddress.Parse("FF1E:03AD::7F2E:172A:1E24")), "The media RTP address was not correct.");
Assert.True(audioRtpEndPoint.Port == 6000, "The media RTP port was not correct.");
}
public void GetFirstMedaiOfferRTPSocketUnitTest()
{
Logger.LogDebug(System.Reflection.MethodBase.GetCurrentMethod().Name);
string sdpStr =
"v=0" + m_CRLF +
"o=root 3285 3285 IN IP4 10.0.0.4" + m_CRLF +
"s=session" + m_CRLF +
"c=IN IP4 10.0.0.4" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 12228 RTP/AVP 0 101" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=silenceSupp:off - - - -" + m_CRLF +
"a=ptime:20" + m_CRLF +
"a=sendrecv";
IPEndPoint audioRtpEndPoint = SDP.GetSDPRTPEndPoint(sdpStr);
Assert.True(audioRtpEndPoint.Address.Equals(IPAddress.Parse("10.0.0.4")), "The media RTP address was not correct.");
Assert.True(audioRtpEndPoint.Port == 12228, "The media RTP port was not correct.");
}
private static readonly int RTP_REPORTING_PERIOD_SECONDS = 5; // Period at which to write RTP stats.
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
IPAddress defaultAddr = LocalIPConfig.GetDefaultIPv4Address();
var sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new SIPTransactionEngine());
int port = FreePort.FindNextAvailableUDPPort(SIPConstants.DEFAULT_SIP_PORT);
var sipChannel = new SIPUDPChannel(new IPEndPoint(defaultAddr, port));
sipTransport.AddSIPChannel(sipChannel);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource uasCts = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept a little it simpler.
uas?.Hangup();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uasCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(defaultAddr, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, uasCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup();
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup();
uasCts?.Cancel();
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
if (uas?.IsHungup == false)
{
uas?.Hangup();
}
uasCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
private static 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(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2("localhost.pfx");
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
//sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
RtpAVSession rtpSession = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)SDPMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
rtpSession = new RtpAVSession(
new AudioOptions {
AudioSource = AudioSourcesEnum.Music, SourceFile = executableDir + "/" + AUDIO_FILE_PCMU
}, null);
rtpSession.setRemoteDescription(new RTCSessionDescription {
type = RTCSdpType.offer, sdp = offerSdp
});
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
await sipTransport.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
rtpCts = new CancellationTokenSource();
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.CloseSession(null);
};
rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
var answerSdp = await rtpSession.createAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpSession?.CloseSession(null);
rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
public void Call(string destination)
{
CreateMediaSockets(SDP.GetSDPRTPEndPoint(m_uas.CallRequest.Body));
m_xmppCall.PlaceCall(destination + "@" + GOOGLE_VOICE_HOST, GetSDPForXMPPRequest());
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"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 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
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent server example.");
Console.WriteLine("Press ctrl-c to exit.");
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.ContactHost = Dns.GetHostName();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
//sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
//if (File.Exists("localhost.pfx"))
//{
// var certificate = new X509Certificate2(@"localhost.pfx", "");
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.Any, SIPS_LISTEN_PORT)));
// sipTransport.AddSIPChannel(new SIPTLSChannel(certificate, new IPEndPoint(IPAddress.IPv6Any, SIPS_LISTEN_PORT)));
//}
//EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
// In practice there could be a number of reasons to reject the call. Unsupported extensions, mo matching codecs etc. etc.
if (sipRequest.Header.HasUnknownRequireExtension)
{
// The caller requires an extension that we don't support.
SIPSorcery.Sys.Log.Logger.LogWarning($"Rejecting incoming call due to one or more required exensions not being supported, required extensions: {sipRequest.Header.Require}.");
uas.Reject(SIPResponseStatusCodesEnum.NotImplemented, "Unsupported Require Extension", null);
}
else
{
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Simulating answer delay to test provisional response retransmits.
await Task.Delay(2000);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localSIPEndPoint.Address, 49000, 49100, false, out rtpSocket, out controlSocket);
IPEndPoint rtpEndPoint = rtpSocket.LocalEndPoint as IPEndPoint;
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
var rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, AUDIO_FILE, rtpCts))
.ContinueWith(_ => { if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS)
{
try
{
SIPSorcery.Sys.Log.Logger.LogInformation($"{localSIPEndPoint.ToString()}<-{remoteEndPoint.ToString()}: {sipRequest.StatusLine}");
//SIPSorcery.Sys.Log.Logger.LogDebug(sipRequest.ToString());
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
catch (Exception optionsExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Failed to send SIP OPTIONS response. {optionsExcp.Message}");
}
}
};
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
rtpCts?.Cancel();
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
SIPSorcery.Sys.Log.Logger.LogInformation("Waiting 1s for call to hangup...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
};
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery user agent server example.");
Console.WriteLine("Press h to hangup a call or ctrl-c to exit.");
EnableConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogWarning($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var localhostCertificate = new X509Certificate2("localhost.pfx");
// IPv4 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenAddress, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
// IPv6 channels.
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenIPv6Address, SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTLSChannel(localhostCertificate, new IPEndPoint(listenIPv6Address, SIPS_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_SECURE_WEBSOCKET_LISTEN_PORT, localhostCertificate));
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.G722)))
{
Log.LogDebug($"Using G722 RTP media type and audio file {AUDIO_FILE_G722}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.G722, null, null);
audioFile = AUDIO_FILE_G722;
}
else if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, null, null, SIPCallDirection.In, null, null, null, uasTransaction);
rtpCts = new CancellationTokenSource();
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(dstRtpEndPoint.AddressFamily == AddressFamily.InterNetworkV6 ? IPAddress.IPv6Any : IPAddress.Any, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
IPEndPoint rtpEndPoint = new IPEndPoint(rtpAddress, (rtpSocket.LocalEndPoint as IPEndPoint).Port);
var rtpTask = Task.Run(() => SendRecvRtp(rtpSocket, rtpSession, dstRtpEndPoint, audioFile, rtpCts))
.ContinueWith(_ =>
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
});
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, GetSDP(rtpEndPoint).ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Call hungup.");
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
SIPSorcery.Sys.Log.Logger.LogInformation("Exiting...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
Hangup(uas).Wait();
rtpCts?.Cancel();
rtpSocket?.Close();
controlSocket?.Close();
}
if (keyProps.KeyChar == 'q')
{
SIPSorcery.Sys.Log.Logger.LogInformation("Quitting...");
if (sipTransport != null)
{
SIPSorcery.Sys.Log.Logger.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
SIPSorcery.Sys.Log.Logger.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
private static ConcurrentQueue <RTPEvent> _dtmfEvents = new ConcurrentQueue <RTPEvent>(); // Add a DTMF event to this queue to have the it sent
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"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();
}
}
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();
}
};
}
private static void SIPTransportRequestReceived(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
var rtpJob = (from job in m_rtpJobs.Values where job.UAS.CallRequest.Header.CallId == sipRequest.Header.CallId select job).FirstOrDefault();
if (rtpJob != null)
{
rtpJob.Stop();
// Call has been hungup by remote end.
Console.WriteLine("Call hungup by client: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
Publish(rtpJob.QueueName, "BYE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
//Console.WriteLine("Request Received " + localSIPEndPoint + "<-" + remoteEndPoint + "\n" + sipRequest.ToString());
//m_uas.SIPDialogue.Hangup(m_sipTransport, null);
SIPResponse okResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
m_sipTransport.SendResponse(okResponse);
}
else
{
Console.WriteLine("Unmatched BYE request received for " + sipRequest.URI.ToString() + ".\n");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Console.WriteLine("Incoming call request: " + localSIPEndPoint + "<-" + remoteEndPoint + " " + sipRequest.URI.ToString() + ".\n");
Publish(sipRequest.URI.User, "INVITE request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
Console.WriteLine(sipRequest.Body);
SIPPacketMangler.MangleSIPRequest(SIPMonitorServerTypesEnum.Unknown, sipRequest, null, LogTraceMessage);
UASInviteTransaction uasTransaction = m_sipTransport.CreateUASTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
var uas = new SIPServerUserAgent(m_sipTransport, null, null, null, SIPCallDirection.In, null, null, LogTraceMessage, uasTransaction);
uas.CallCancelled += UASCallCancelled;
RTPDiagnosticsJob rtpJob = new RTPDiagnosticsJob(m_rtpListenIPAddress, m_publicIPAddress, uas, sipRequest);
string sdpAddress = SDP.GetSDPRTPEndPoint(sipRequest.Body).Address.ToString();
// Only mangle if there is something to change. For example the server could be on the same private subnet in which case it can't help.
IPEndPoint expectedRTPEndPoint = new IPEndPoint(rtpJob.RemoteRTPEndPoint.Address, rtpJob.RemoteRTPEndPoint.Port);
if (IPSocket.IsPrivateAddress(rtpJob.RemoteRTPEndPoint.Address.ToString()))
{
expectedRTPEndPoint.Address = remoteEndPoint.Address;
}
Publish(sipRequest.URI.User, "Advertised RTP remote socket " + rtpJob.RemoteRTPEndPoint + ", expecting from " + expectedRTPEndPoint + ".");
m_rtpJobs.Add(rtpJob.RTPListenEndPoint.Port, rtpJob);
//ThreadPool.QueueUserWorkItem(delegate { StartRTPListener(rtpJob); });
Console.WriteLine(rtpJob.LocalSDP.ToString());
uas.Answer("application/sdp", rtpJob.LocalSDP.ToString(), CallProperties.CreateNewTag(), null, SIPDialogueTransferModesEnum.NotAllowed);
var hangupTimer = new Timer(delegate
{
if (!rtpJob.StopJob)
{
if (uas != null && uas.SIPDialogue != null)
{
if (rtpJob.RTPPacketReceived && !rtpJob.ErrorOnRTPSend)
{
Publish(sipRequest.URI.User, "Test completed. There were no RTP send or receive errors.");
}
else if (!rtpJob.RTPPacketReceived)
{
Publish(sipRequest.URI.User, "Test completed. An error was identified, no RTP packets were received.");
}
else
{
Publish(sipRequest.URI.User, "Test completed. An error was identified, there was a problem when attempting to send an RTP packet.");
}
rtpJob.Stop();
uas.SIPDialogue.Hangup(m_sipTransport, null);
}
}
}, null, HANGUP_TIMEOUT, Timeout.Infinite);
}
else if (sipRequest.Method == SIPMethodsEnum.CANCEL)
{
UASInviteTransaction inviteTransaction = (UASInviteTransaction)m_sipTransport.GetTransaction(SIPTransaction.GetRequestTransactionId(sipRequest.Header.Vias.TopViaHeader.Branch, SIPMethodsEnum.INVITE));
if (inviteTransaction != null)
{
Console.WriteLine("Matching CANCEL request received " + sipRequest.URI.ToString() + ".\n");
Publish(sipRequest.URI.User, "CANCEL request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
SIPCancelTransaction cancelTransaction = m_sipTransport.CreateCancelTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, inviteTransaction);
cancelTransaction.GotRequest(localSIPEndPoint, remoteEndPoint, sipRequest);
}
else
{
Console.WriteLine("No matching transaction was found for CANCEL to " + sipRequest.URI.ToString() + ".\n");
SIPResponse noCallLegResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
m_sipTransport.SendResponse(noCallLegResponse);
}
}
else
{
Console.WriteLine("SIP " + sipRequest.Method + " request received but no processing has been set up for it, rejecting.\n");
Publish(sipRequest.URI.User, sipRequest.Method + " request received from " + remoteEndPoint + " for " + sipRequest.URI.ToString() + ".");
SIPResponse notAllowedResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
m_sipTransport.SendResponse(notAllowedResponse);
}
}
static void Main()
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP transport and RTP stream.
AddConsoleLogger();
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
// Un/comment this line to see/hide each SIP message sent and received.
EnableTraceLogs(sipTransport);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
SIPUserAgent userAgent = new SIPUserAgent(sipTransport, null);
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
Socket rtpSocket = null;
Socket controlSocket = null;
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
userAgent.InDialogRequestReceivedAsync(sipRequest).Wait();
}
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
SIPSorcery.Sys.Log.Logger.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
// Check there's a codec we support in the INVITE offer.
var offerSdp = SDP.ParseSDPDescription(sipRequest.Body);
IPEndPoint dstRtpEndPoint = SDP.GetSDPRTPEndPoint(sipRequest.Body);
RTPSession rtpSession = null;
string audioFile = null;
if (offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.HasMediaFormat((int)RTPPayloadTypesEnum.PCMU)))
{
Log.LogDebug($"Using PCMU RTP media type and audio file {AUDIO_FILE_PCMU}.");
rtpSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
audioFile = AUDIO_FILE_PCMU;
}
if (rtpSession == null)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, null);
sipTransport.SendResponse(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (userAgent?.IsAnswered == true)
{
userAgent?.Hangup();
}
rtpCts?.Cancel();
UASInviteTransaction uasTransaction = sipTransport.CreateUASTransaction(sipRequest, null);
if (userAgent.AcceptCall(uasTransaction))
{
rtpCts = new CancellationTokenSource();
// The RTP socket is listening on IPAddress.Any but the IP address placed into the SDP needs to be one the caller can reach.
IPAddress rtpAddress = NetServices.GetLocalAddressForRemote(dstRtpEndPoint.Address);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
NetServices.CreateRtpSocket(rtpAddress, RTP_PORT_START, RTP_PORT_END, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
rtpSendSession.DestinationEndPoint = dstRtpEndPoint;
rtpRecvSession.OnReceiveFromEndPointChanged += (oldEP, newEP) =>
{
Log.LogDebug($"RTP destination end point changed from {oldEP} to {newEP}.");
rtpSendSession.DestinationEndPoint = newEP;
};
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
userAgent.Answer(GetSDP(rtpSocket.LocalEndPoint as IPEndPoint));
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
sipTransport.SendResponse(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
sipTransport.SendResponse(optionsResponse);
}
}
catch (Exception reqExcp)
{
SIPSorcery.Sys.Log.Logger.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitCts.Cancel();
rtpCts?.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler.
Task.Run(async() =>
{
try
{
while (!exitCts.Token.WaitHandle.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 't')
{
// Initiate a transfer.
bool transferResult = await userAgent.Transfer(SIPURI.ParseSIPURI(TRANSFER_DESTINATION_SIP_URI), new TimeSpan(0, 0, TRANSFER_TIMEOUT_SECONDS), exitCts.Token);
if (transferResult)
{
// If the transfer was accepted the original call will already have been hungup.
userAgent = null;
exitCts.Cancel();
}
else
{
Log.LogWarning($"Transfer to {TRANSFER_DESTINATION_SIP_URI} failed.");
}
}
else if (keyProps.KeyChar == 'q')
{
// Quit application.
exitCts.Cancel();
}
}
}
catch (Exception excp)
{
Log.LogError($"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 (userAgent != null)
{
if (userAgent.IsAnswered)
{
Log.LogInformation($"Hanging up call to {userAgent?.CallDescriptor?.To}.");
userAgent.Hangup();
}
// Give the final request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main()
{
Console.WriteLine("SIPSorcery call hold example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource exitCts = new CancellationTokenSource(); // Cancellation token to stop the SIP trnasport and RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
Log.LogInformation($"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();
}
}
