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 treat our RTP stream as inactive.
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
var localOfferAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
var audioOptions = new AudioOptions
{
AudioSource = AudioSourcesEnum.Microphone,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.PCMU
},
OutputDeviceIndex = AudioOptions.DEFAULT_OUTPUTDEVICE_INDEX
};
var rtpSession = new RtpAVSession(audioOptions, null);
var offerSDP = rtpSession.CreateOffer(localOfferAddress);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
if (resp.Status == SIPResponseStatusCodesEnum.SessionProgress)
{
rtpSession.Start();
}
};
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (iuac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var result = rtpSession.SetRemoteDescription(SDP.ParseSDPDescription(resp.Body));
if (result == SetDescriptionResultEnum.OK)
{
rtpSession.Start();
}
else
{
Log.LogWarning($"Failed to set remote description {result}.");
uac.Hangup();
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(okResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
exitMre.Set();
}
}
};
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
offerSDP.ToString(),
null);
uac.Call(callDescriptor);
uac.ServerTransaction.TransactionTraceMessage += (tx, msg) => Log.LogInformation($"UAC tx trace message. {msg}");
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
rtpSession.CloseSession(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
public static string Validate(SIPProvider sipProvider)
{
TypeDescriptor.AddProviderTransparent(new AssociatedMetadataTypeTypeDescriptionProvider(typeof(SIPProvider), typeof(SIPProviderMetadata)), typeof(SIPProvider));
var validationContext = new ValidationContext(sipProvider, null, null);
var validationResults = new List <ValidationResult>();
Validator.TryValidateObject(sipProvider, validationContext, validationResults);
if (validationResults.Count > 0)
{
return(validationResults.First().ErrorMessage);
}
else
{
Guid testGuid = Guid.Empty;
if (!Guid.TryParse(sipProvider.ID, out testGuid))
{
return("The ID was not a valid GUID.");
}
if (sipProvider.ProviderType == ProviderTypes.SIP.ToString())
{
if (sipProvider.ProviderServer.IsNullOrBlank())
{
return("A value for Server must be specified.");
}
if (sipProvider.RegisterEnabled && sipProvider.RegisterContact == null)
{
return("A valid contact must be supplied to enable a provider registration.");
}
//else if (sipProvider.RegisterServer != null && sipProvider.m_registerServer.Host.IndexOf('.') == -1)
//{
// return "Your register server entry appears to be invalid. A valid hostname or IP address should contain at least one '.'.";
//}
//else if (sipProvider.RegisterContact != null && sipProvider.m_registerContact.Host.IndexOf('.') == -1)
//{
// return "Your register contact entry appears to be invalid. A valid hostname or IP address should contain at least one '.'.";
//}
//else if (sipProvider.RegisterContact != null && sipProvider.RegisterContact.User.IsNullOrBlank())
//{
// return "Your register contact entry appears to be invalid, the user portion was missing. Contacts must be of the form [email protected], e.g. [email protected].";
//}
else if (ProhibitedServerPatterns != null && Regex.Match(sipProvider.ProviderServer, ProhibitedServerPatterns).Success)
{
return("The Provider Server contains a disallowed string. If you are trying to create a Provider entry pointing to sipsorcery.com it is not permitted.");
}
else if (ProhibitedServerPatterns != null && sipProvider.RegisterServer != null && Regex.Match(sipProvider.RegisterServer, ProhibitedServerPatterns).Success)
{
return("The Provider Register Server contains a disallowed string. If you are trying to create a Provider entry pointing to sipsorcery.com it is not permitted.");
}
else if (!SIPURI.TryParse(sipProvider.ProviderServer))
{
return("The Provider Server could not be parsed as a valid SIP URI.");
}
else if (sipProvider.RegisterServer != null && !SIPURI.TryParse(sipProvider.RegisterServer))
{
return("The Register Server could not be parsed as a valid SIP URI.");
}
}
else if (sipProvider.ProviderType == ProviderTypes.GoogleVoice.ToString())
{
if (sipProvider.ProviderPassword.IsNullOrBlank())
{
return("A password is required for Google Voice entries.");
}
else if (sipProvider.GVCallbackNumber.IsNullOrBlank())
{
return("A callback number is required for Google Voice entries.");
}
else if (Regex.Match(sipProvider.GVCallbackNumber, @"\D").Success)
{
return("The callback number contains an invalid character. Only digits are permitted.");
}
}
}
return(null);
}
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 preferIPv6 = false;
bool isCallHungup = false;
bool hasCallFailed = false;
Log = AddConsoleLogger(LogEventLevel.Verbose);
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args?.Length > 0)
{
if (!SIPURI.TryParse(args[0], out callUri))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
}
if (args?.Length > 1 && args[1] == "ipv6")
{
preferIPv6 = true;
}
if (preferIPv6)
{
Log.LogInformation($"Call destination {callUri}, preferencing IPv6.");
}
else
{
Log.LogInformation($"Call destination {callUri}.");
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
sipTransport.PreferIPv6NameResolution = preferIPv6;
sipTransport.EnableTraceLogs();
var audioSession = new WindowsAudioEndPoint(new AudioEncoder());
audioSession.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMA || x.Codec == AudioCodecsEnum.PCMU);
//audioSession.RestrictFormats(x => x.Codec == AudioCodecsEnum.G722);
var rtpSession = new VoIPMediaSession(audioSession.ToMediaEndPoints());
var offerSDP = rtpSession.CreateOffer(preferIPv6 ? IPAddress.IPv6Any : IPAddress.Any);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallRinging += async(uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
if (resp.Status == SIPResponseStatusCodesEnum.SessionProgress)
{
if (resp.Body != null)
{
var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body));
if (result == SetDescriptionResultEnum.OK)
{
await rtpSession.Start();
Log.LogInformation($"Remote SDP set from in progress response. RTP session started.");
}
}
}
};
uac.CallFailed += (uac, err, resp) =>
{
Log.LogWarning($"Call attempt to {uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += async(iuac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
if (resp.Body != null)
{
var result = rtpSession.SetRemoteDescription(SdpType.answer, SDP.ParseSDPDescription(resp.Body));
if (result == SetDescriptionResultEnum.OK)
{
await rtpSession.Start();
}
else
{
Log.LogWarning($"Failed to set remote description {result}.");
uac.Hangup();
}
}
else if (!rtpSession.IsStarted)
{
Log.LogWarning($"Failed to set get remote description in session progress or final response.");
uac.Hangup();
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += async(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse okResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(okResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
exitMre.Set();
}
}
};
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
offerSDP.ToString(),
null);
uac.Call(callDescriptor, null);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
exitMre.Set();
};
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
exitMre.WaitOne();
Log.LogInformation("Exiting...");
rtpSession.Close(null);
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
private static int INPUT_SAMPLE_PERIOD_MILLISECONDS = 20; // This sets the frequency of the RTP packets.
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
ManualResetEvent exitMre = new ManualResetEvent(false);
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0], out callUri))
{
Log.LogWarning($"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($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
AddConsoleLogger();
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
else
{
callUri = SIPURI.ParseSIPURIRelaxed(args[0]);
}
}
Log.LogInformation($"Call destination {callUri}.");
var lookupResult = SIPDNSManager.ResolveSIPService(callUri, false);
Log.LogDebug($"DNS lookup result for {callUri}: {lookupResult?.GetSIPEndPoint()}.");
var dstAddress = lookupResult.GetSIPEndPoint().Address;
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
var listenAddress = dstAddress.AddressFamily == AddressFamily.InterNetwork ? IPAddress.Any : IPAddress.IPv6Any;
if (callUri.Scheme == SIPSchemesEnum.sip && callUri.Protocol == SIPProtocolsEnum.udp)
{
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(listenAddress, 0)));
}
else if (callUri.Scheme == SIPSchemesEnum.sip && callUri.Protocol == SIPProtocolsEnum.tcp)
{
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(listenAddress, 0)));
}
else if (callUri.Scheme == SIPSchemesEnum.sips || callUri.Protocol == SIPProtocolsEnum.tls)
{
sipTransport.AddSIPChannel(new SIPTLSChannel(new X509Certificate2("localhost.pfx"), new IPEndPoint(listenAddress, 0)));
}
EnableTraceLogs(sipTransport);
IPAddress localIPAddress = NetServices.GetLocalAddressForRemote(dstAddress);
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
NetServices.CreateRtpSocket(localIPAddress, 48000, 48100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
// Only set the remote RTP end point if there hasn't already been a packet received on it.
if (_remoteRtpEndPoint == null)
{
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
callUri.CanonicalAddress,
null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
rtpCts.Token.WaitHandle.WaitOne();
Log.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
static void Main(string[] args)
{
Console.WriteLine("SIPSorcery client user agent example.");
Console.WriteLine("Press ctrl-c to exit.");
// Plumbing code to facilitate a graceful exit.
CancellationTokenSource rtpCts = new CancellationTokenSource(); // Cancellation token to stop the RTP stream.
bool isCallHungup = false;
bool hasCallFailed = false;
// Logging configuration. Can be ommitted if internal SIPSorcery debug and warning messages are not required.
var loggerFactory = new Microsoft.Extensions.Logging.LoggerFactory();
var loggerConfig = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
loggerFactory.AddSerilog(loggerConfig);
SIPSorcery.Sys.Log.LoggerFactory = loggerFactory;
SIPURI callUri = SIPURI.ParseSIPURI(DEFAULT_DESTINATION_SIP_URI);
if (args != null && args.Length > 0)
{
if (!SIPURI.TryParse(args[0]))
{
Log.LogWarning($"Command line argument could not be parsed as a SIP URI {args[0]}");
}
else
{
callUri = SIPURI.ParseSIPURIRelaxed(args[0]);
}
}
Log.LogInformation($"Call destination {callUri}.");
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
int port = SIPConstants.DEFAULT_SIP_PORT + 1000;
IPAddress localAddress = sipTransport.GetLocalAddress(IPAddress.Parse("8.8.8.8"));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(localAddress, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, port)));
//sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, port)));
//EnableTraceLogs(sipTransport);
// Select the IP address to use for RTP based on the destination SIP URI.
var endPointForCall = callUri.ToSIPEndPoint() == null ? sipTransport.GetDefaultSIPEndPoint(callUri.Protocol) : sipTransport.GetDefaultSIPEndPoint(callUri.ToSIPEndPoint());
// Initialise an RTP session to receive the RTP packets from the remote SIP server.
Socket rtpSocket = null;
Socket controlSocket = null;
// TODO (find something better): If the SIP endpoint is using 0.0.0.0 for SIP use loopback for RTP.
IPAddress rtpAddress = localAddress;
NetServices.CreateRtpSocket(rtpAddress, 49000, 49100, false, out rtpSocket, out controlSocket);
var rtpRecvSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
var rtpSendSession = new RTPSession((int)RTPPayloadTypesEnum.PCMU, null, null);
// Create a client user agent to place a call to a remote SIP server along with event handlers for the different stages of the call.
var uac = new SIPClientUserAgent(sipTransport);
uac.CallTrying += (uac, resp) =>
{
Log.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
};
uac.CallRinging += (uac, resp) => Log.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
uac.CallFailed += (uac, err) =>
{
Log.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
hasCallFailed = true;
};
uac.CallAnswered += (uac, resp) =>
{
if (resp.Status == SIPResponseStatusCodesEnum.Ok)
{
Log.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
_remoteRtpEndPoint = SDP.GetSDPRTPEndPoint(resp.Body);
Log.LogDebug($"Remote RTP socket {_remoteRtpEndPoint}.");
}
else
{
Log.LogWarning($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
}
};
// The only incoming request that needs to be explicitly handled for this example is if the remote end hangs up the call.
sipTransport.SIPTransportRequestReceived += (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPNonInviteTransaction byeTransaction = sipTransport.CreateNonInviteTransaction(sipRequest, remoteEndPoint, localSIPEndPoint, null);
SIPResponse byeResponse = SIPTransport.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
byeTransaction.SendFinalResponse(byeResponse);
if (uac.IsUACAnswered)
{
Log.LogInformation("Call was hungup by remote server.");
isCallHungup = true;
rtpCts.Cancel();
}
}
};
// It's a good idea to start the RTP receiving socket before the call request is sent.
// A SIP server will generally start sending RTP as soon as it has processed the incoming call request and
// being ready to receive will stop any ICMP error response being generated.
Task.Run(() => RecvRtp(rtpSocket, rtpRecvSession, rtpCts));
Task.Run(() => SendRtp(rtpSocket, rtpSendSession, rtpCts));
// Start the thread that places the call.
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
SIPConstants.SIP_DEFAULT_USERNAME,
null,
callUri.ToString(),
SIPConstants.SIP_DEFAULT_FROMURI,
null, null, null, null,
SIPCallDirection.Out,
SDP.SDP_MIME_CONTENTTYPE,
GetSDP(rtpSocket.LocalEndPoint as IPEndPoint).ToString(),
null);
uac.Call(callDescriptor);
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
rtpCts.Cancel();
};
// At this point the call has been initiated and everything will be handled in an event handler or on the RTP
// receive task. The code below is to gracefully exit.
// Wait for a signal saying the call failed, was cancelled with ctrl-c or completed.
rtpCts.Token.WaitHandle.WaitOne();
Log.LogInformation("Exiting...");
rtpSocket?.Close();
controlSocket?.Close();
if (!isCallHungup && uac != null)
{
if (uac.IsUACAnswered)
{
Log.LogInformation($"Hanging up call to {uac.CallDescriptor.To}.");
uac.Hangup();
}
else if (!hasCallFailed)
{
Log.LogInformation($"Cancelling call to {uac.CallDescriptor.To}.");
uac.Cancel();
}
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to clean up...");
Task.Delay(1000).Wait();
}
SIPSorcery.Net.DNSManager.Stop();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
}
