static void Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
Log = AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
_sipTransport = new SIPTransport();
_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
var userAgent = new SIPUserAgent(_sipTransport, null, true);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
userAgent.OnIncomingCall += async(ua, req) =>
{
WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession voipMediaSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;
var uas = userAgent.AcceptCall(req);
await userAgent.Answer(uas, voipMediaSession);
};
Console.WriteLine("press any key to exit...");
Console.Read();
// Clean up.
_sipTransport.Shutdown();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSink = winAudio, AudioSource = winAudio
});
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new WindowsAudioRtpSession();
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
Task.Delay(1000).Wait();
}
// Clean up.
sipTransport.Shutdown();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started PortAudio Demo (YMMV)");
AddConsoleLogger();
var userAgent = new SIPUserAgent();
var portAudioEndPoint = new PortAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(portAudioEndPoint.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
await Task.Delay(1000);
}
}
public async Task IncomingCallNoSdpUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 0" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF + m_CRLF;
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Loopback, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
await userAgent.Answer(uas, CreateMockVoIPMediaEndPoint());
// The call attempt should timeout while waiting for the ACK request with the SDP answer.
Assert.False(userAgent.IsCallActive);
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The suer agent the RTP session is being created for.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua)
{
var rtpAudioSession = new RtpAudioSession(AddressFamily.InterNetwork);
// Add the required audio capabilities to the RTP session. These will
// automatically get used when creating SDP offers/answers.
var pcma = new SDPMediaFormat(SDPMediaFormatsEnum.PCMA);
// RTP event support.
int clockRate = pcma.GetClockRate();
SDPMediaFormat rtpEventFormat = new SDPMediaFormat(DTMF_EVENT_PAYLOAD_ID);
rtpEventFormat.SetFormatAttribute($"{RTPSession.TELEPHONE_EVENT_ATTRIBUTE}/{clockRate}");
rtpEventFormat.SetFormatParameterAttribute("0-16");
var audioCapabilities = new List <SDPMediaFormat> {
pcma, rtpEventFormat
};
MediaStreamTrack audioTrack = new MediaStreamTrack(null, SDPMediaTypesEnum.audio, false, audioCapabilities);
rtpAudioSession.addTrack(audioTrack);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (type, rtp) => OnRtpPacketReceived(ua, type, rtp);
if (_sendSilenceTimer == null)
{
_sendSilenceTimer = new Timer(SendSilence, null, 0, SEND_SILENCE_PERIOD_MS);
}
return(rtpAudioSession);
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new RtpAVSession(new AudioOptions {
AudioSource = AudioSourcesEnum.Microphone
}, null);
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua)
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}
public async Task CancelCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport aliceTransport = new SIPTransport();
aliceTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var alice = new SIPUserAgent(aliceTransport, null, true);
SIPServerUserAgent uas = null;
// Auto accept but NOT answering.
alice.OnIncomingCall += (ua, req) => uas = ua.AcceptCall(req);
SIPTransport bobTransport = new SIPTransport();
bobTransport.AddSIPChannel(new SIPUDPChannel(IPAddress.Loopback, 0));
var bob = new SIPUserAgent(bobTransport, null, true);
var callTask = bob.Call(alice.ContactURI.ToString(), null, null, CreateMediaSession());
await Task.Delay(500);
Assert.True(bob.IsRinging);
Assert.NotNull(uas);
Assert.False(uas.IsCancelled);
bob.Cancel();
await Task.Delay(500);
Assert.False(alice.IsCallActive);
Assert.False(bob.IsCallActive);
Assert.True(uas.IsCancelled);
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
var userAgent = new SIPUserAgent(_sipTransport, null);
userAgent.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var uas = userAgent.AcceptCall(sipRequest);
await userAgent.Answer(uas, rtpSession);
if (userAgent.IsCallActive)
{
await rtpSession.Start();
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
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)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Header.From != null &&
sipRequest.Header.From.FromTag != null &&
sipRequest.Header.To != null &&
sipRequest.Header.To.ToTag != null)
{
// This is an in-dialog request that will be handled directly by a user agent instance.
}
else if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
ua.OnCallHungup += OnHangup;
ua.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
ua.OnRtpEvent += (evt, hdr) => Log.LogDebug($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
ua.OnTransactionTraceMessage += (tx, msg) => Log.LogDebug($"uas tx {tx.TransactionId}: {msg}");
ua.ServerCallRingTimeout += (uas) =>
{
Log.LogWarning($"Incoming call timed out in {uas.ClientTransaction.TransactionState} state waiting for client ACK, terminating.");
ua.Hangup();
};
var uas = ua.AcceptCall(sipRequest);
var rtpSession = CreateRtpSession(ua, sipRequest.URI.User);
await ua.Answer(uas, rtpSession);
if (ua.IsCallActive)
{
await rtpSession.Start();
_calls.TryAdd(ua.Dialogue.CallId, ua);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Getting Started Demo");
AddConsoleLogger();
_waveFile = new WaveFileWriter("output.mp3", _waveFormat);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallFailed += (uac, err, resp) =>
{
Console.WriteLine($"Call failed {err}");
_waveFile?.Close();
};
userAgent.OnCallHungup += (dialog) => _waveFile?.Close();
WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
VoIPMediaSession voipSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
voipSession.OnRtpPacketReceived += OnRtpPacketReceived;
// Ctrl-c will gracefully exit the call at any point.
Console.CancelKeyPress += delegate(object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
else
{
Console.WriteLine("Cancelling call");
userAgent.Cancel();
}
};
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioSource = AudioSourcesEnum.SineWave;
if (string.IsNullOrEmpty(dst) || !Enum.TryParse <AudioSourcesEnum>(dst, out audioSource))
{
audioSource = AudioSourcesEnum.Silence;
}
var audioOptions = new AudioSourceOptions {
AudioSource = audioSource
};
if (audioSource == AudioSourcesEnum.Music)
{
audioOptions.SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>();
if (codecs.Contains(SDPMediaFormatsEnum.PCMA))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.PCMA, MUSIC_FILE_PCMA);
}
if (codecs.Contains(SDPMediaFormatsEnum.PCMU))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.PCMU, MUSIC_FILE_PCMU);
}
if (codecs.Contains(SDPMediaFormatsEnum.G722))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.G722, MUSIC_FILE_G722);
}
}
;
Log.LogInformation($"RTP audio session source set to {audioSource}.");
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}
static void Main()
{
Console.WriteLine("SIPSorcery sip.js Demo");
Log = AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var sipChannel = new SIPWebSocketChannel(IPAddress.Loopback, 8081);
sipTransport.AddSIPChannel(sipChannel);
var userAgent = new SIPUserAgent(sipTransport, null, true);
userAgent.OnIncomingCall += async(ua, req) =>
{
Log.LogDebug($"Auto-answering incoming call from {req.Header.From}.");
var uas = userAgent.AcceptCall(req);
var peerConnection = new RTCPeerConnection(null);
peerConnection.onconnectionstatechange += (state) =>
{
Log.LogDebug($"Peer connection state change to {state}.");
if (state == RTCPeerConnectionState.failed)
{
peerConnection.Close("ice disconnection");
}
else if (state == RTCPeerConnectionState.connected)
{
peerConnection.OnRtpPacketReceived += OnRtpPacketReceived;
}
else if (state == RTCPeerConnectionState.closed)
{
peerConnection.OnRtpPacketReceived -= OnRtpPacketReceived;
}
};
MediaStreamTrack audioTrack = new MediaStreamTrack(new List <AudioFormat> {
new AudioFormat(SDPWellKnownMediaFormatsEnum.PCMU)
}, MediaStreamStatusEnum.SendRecv);
peerConnection.addTrack(audioTrack);
//MediaStreamTrack videoTrack = new MediaStreamTrack("1", SDPMediaTypesEnum.video, false, new List<SDPMediaFormat> { new SDPMediaFormat(SDPMediaFormatsEnum.VP8) }, MediaStreamStatusEnum.Inactive);
//peerConnection.addTrack(videoTrack);
var answerResult = await userAgent.Answer(uas, peerConnection);
};
Console.Write("press any key to exit...");
Console.Read();
sipTransport.Shutdown();
}
/// <summary>
/// An asynchronous task that attempts to initiate a new call to a listening UAS.
/// </summary>
/// <param name="sipTransport">The transport object to use for the send.</param>
/// <param name="dst">The destination end point to send the request to.</param>
/// <returns>True if the expected response was received, false otherwise.</returns>
private static async Task <bool> InitiateCallTaskAsync(SIPTransport sipTransport, SIPURI dst)
{
//UdpClient hepClient = new UdpClient(0, AddressFamily.InterNetwork);
try
{
//sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
//{
// logger.LogDebug($"Request sent: {localEP}->{remoteEP}");
// logger.LogDebug(req.ToString());
// //var hepBuffer = HepPacket.GetBytes(localEP, remoteEP, DateTimeOffset.Now, 333, "myHep", req.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
//sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
//{
// logger.LogDebug($"Response received: {localEP}<-{remoteEP}");
// logger.LogDebug(resp.ToString());
// //var hepBuffer = HepPacket.GetBytes(remoteEP, localEP, DateTimeOffset.Now, 333, "myHep", resp.ToString());
// //hepClient.SendAsync(hepBuffer, hepBuffer.Length, "192.168.11.49", 9060);
//};
var ua = new SIPUserAgent(sipTransport, null);
ua.ClientCallTrying += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallRinging += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
ua.ClientCallFailed += (uac, err, resp) => logger.LogWarning($"{uac.CallDescriptor.To} Failed: {err}");
ua.ClientCallAnswered += (uac, resp) => logger.LogInformation($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictCodecs(new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU
});
var voipMediaSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
var result = await ua.Call(dst.ToString(), null, null, voipMediaSession);
ua.Hangup();
await Task.Delay(200);
return(result);
}
catch (Exception excp)
{
logger.LogError($"Exception InitiateCallTaskAsync. {excp.Message}");
return(false);
}
}
public async Task BlindTransferCancelUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
transport.AddSIPChannel(new MockSIPChannel(new System.Net.IPEndPoint(IPAddress.Any, 0)));
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = "INVITE sip:192.168.11.50:5060 SIP/2.0" + m_CRLF +
"Via: SIP/2.0/UDP 192.168.11.50:60163;rport;branch=z9hG4bKPj869f70960bdd4204b1352eaf242a3691" + m_CRLF +
"To: <sip:[email protected]>;tag=ZUJSXRRGXQ" + m_CRLF +
"From: <sip:[email protected]>;tag=4a60ce364b774258873ff199e5e39938" + m_CRLF +
"Call-ID: 17324d6df8744d978008c8997bfd208d" + m_CRLF +
"CSeq: 3532 INVITE" + m_CRLF +
"Contact: <sip:[email protected]:60163;ob>" + m_CRLF +
"Max-Forwards: 70" + m_CRLF +
"User-Agent: MicroSIP/3.19.22" + m_CRLF +
"Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS" + m_CRLF +
"Supported: replaces, 100rel, timer, norefersub" + m_CRLF +
"Content-Length: 343" + m_CRLF +
"Content-Type: application/sdp" + m_CRLF +
"Session-Expires: 1800" + m_CRLF +
"Min-SE: 90" + m_CRLF +
"" + m_CRLF +
"v=0" + m_CRLF +
"o=- 3785527268 3785527269 IN IP4 192.168.11.50" + m_CRLF +
"s=pjmedia" + m_CRLF +
"t=0 0" + m_CRLF +
"m=audio 4032 RTP/AVP 0 101" + m_CRLF +
"c=IN IP4 192.168.11.50" + m_CRLF +
"a=rtpmap:0 PCMU/8000" + m_CRLF +
"a=rtpmap:101 telephone-event/8000" + m_CRLF +
"a=fmtp:101 0-16" + m_CRLF +
"a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
UASInviteTransaction uasTx = new UASInviteTransaction(transport, inviteReq, null);
SIPServerUserAgent mockUas = new SIPServerUserAgent(transport, null, null, null, SIPCallDirection.In, null, null, null, uasTx);
await userAgent.Answer(mockUas, CreateMediaSession());
CancellationTokenSource cts = new CancellationTokenSource();
var blindTransferTask = userAgent.BlindTransfer(SIPURI.ParseSIPURIRelaxed("127.0.0.1"), TimeSpan.FromSeconds(2), cts.Token);
cts.Cancel();
Assert.False(await blindTransferTask);
//await Assert.ThrowsAnyAsync<TaskCanceledException>(async () => { bool result = ; });
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Convert Audio");
Log = AddConsoleLogger();
//WaveFormatConversionStream converter = new WaveFormatConversionStream(_format_s16le48k, )
_waveFile = new WaveFileWriter("output_s16le48k.mp3", _format_s16le48k);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) =>
{
Console.WriteLine("Call hungup.");
_waveFile?.Close();
};
//EnableTraceLogs(sipTransport);
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), audioOptions);
audioExtrasSource.RestrictFormats((format) => format.Codec == AudioCodecsEnum.PCMU);
var rtpSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpSession.OnAudioFormatsNegotiated += (formats) =>
{
_ratio = (double)(OUT_SAMPLE_RATE / formats.First().RtpClockRate);
};
rtpSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
}
public async Task PlaceCallUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport serverTransport = new SIPTransport();
SIPUDPChannel udpChannel = new SIPUDPChannel(IPAddress.Loopback, 0);
serverTransport.AddSIPChannel(udpChannel);
// Set up two user agents: one to answer the test call and one to place it.
SIPUserAgent userAgentServer = new SIPUserAgent(serverTransport, null);
SIPUserAgent userAgentClient = new SIPUserAgent(new SIPTransport(), null);
serverTransport.SIPTransportRequestReceived += async(lep, rep, req) =>
{
logger.LogDebug("Request received: " + req.StatusLine);
var uas = userAgentServer.AcceptCall(req);
RtpAudioSession serverAudioSession = new RtpAudioSession(
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
},
new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
});
var answerResult = await userAgentServer.Answer(uas, serverAudioSession);
logger.LogDebug($"Server agent answer result {answerResult}.");
Assert.True(answerResult);
};
var dstUri = udpChannel.GetContactURI(SIPSchemesEnum.sip, new SIPEndPoint(SIPProtocolsEnum.udp, new IPEndPoint(IPAddress.Loopback, 0)));
logger.LogDebug($"Attempting call to {dstUri.ToString()}.");
RtpAudioSession clientAudioSession = new RtpAudioSession(
new AudioSourceOptions {
AudioSource = AudioSourcesEnum.None
},
new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
});
var callResult = await userAgentClient.Call(dstUri.ToString(), null, null, clientAudioSession);
logger.LogDebug($"Client agent answer result {callResult }.");
Assert.True(callResult);
Assert.Equal(SIPDialogueStateEnum.Confirmed, userAgentClient.Dialogue.DialogueState);
Assert.Equal(SIPDialogueStateEnum.Confirmed, userAgentServer.Dialogue.DialogueState);
}
public SIPClient(SIPTransport sipTransport, RTPMediaSessionManager rtpMediaSessionManager)
{
m_sipTransport = sipTransport;
m_rtpMediaSessionManager = rtpMediaSessionManager;
m_userAgent = new SIPUserAgent(m_sipTransport, null);
m_userAgent.ClientCallTrying += CallTrying;
m_userAgent.ClientCallRinging += CallRinging;
m_userAgent.ClientCallAnswered += CallAnswered;
m_userAgent.ClientCallFailed += CallFailed;
m_userAgent.OnCallHungup += CallFinished;
m_userAgent.ServerCallCancelled += IncomingCallCancelled;
m_userAgent.OnTransferNotify += OnTransferNotify;
}
static async Task DialNumber()
{
string fromHeader = (new SIPFromHeader(USERNAME, new SIPURI(USERNAME, DOMAIN, null), null)).ToString();
SIPCallDescriptor callDescriptor = new SIPCallDescriptor(USERNAME, PASSWORD, DEFAULT_CALL_DESTINATION, fromHeader, null, null, null, null, SIPCallDirection.Out, SDP.SDP_MIME_CONTENTTYPE, null, null);
callDescriptor.CallId = "12028883999";
userAgent = new SIPUserAgent(sipTransport, null);
userAgent.ClientCallTrying += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Trying: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallRinging += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Ringing: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.ClientCallFailed += (uac, err, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Failed: {err}, Status code: {resp?.StatusCode}");
userAgent.ClientCallAnswered += (uac, resp) => Console.WriteLine($"{uac.CallDescriptor.To} Answered: {resp.StatusCode} {resp.ReasonPhrase}.");
userAgent.OnDtmfTone += (key, duration) => OnDtmfTone(userAgent, key, duration);
userAgent.OnRtpEvent += (evt, hdr) => Console.WriteLine($"rtp event {evt.EventID}, duration {evt.Duration}, end of event {evt.EndOfEvent}, timestamp {hdr.Timestamp}, marker {hdr.MarkerBit}.");
userAgent.OnCallHungup += OnHangup;
var rtpSession = new RtpAVSession(
new AudioOptions
{
AudioSource = AudioSourcesEnum.CaptureDevice,
AudioCodecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA
}
},
null);
rtpSession.OnRtpPacketReceived += OnRtpPacketReceived;
var callResult = await userAgent.Call(callDescriptor, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
if (callResult)
{
Console.WriteLine("Enter digits one after another");
for (int i = 0; i < 11; i++)
{
var p = Console.ReadLine();
await userAgent.SendDtmf(byte.Parse(p));
}
}
Console.WriteLine("Enter ?");
Console.ReadLine();
await userAgent.SendDtmf(35);
Thread.Sleep(60000);
userAgent.Hangup();
_waveFile.Dispose();
Console.WriteLine("Hangup");
}
public SIPClient(SIPTransport sipTransport)
{
m_sipTransport = sipTransport;
m_userAgent = new SIPUserAgent(m_sipTransport, null);
m_userAgent.ClientCallTrying += CallTrying;
m_userAgent.ClientCallRinging += CallRinging;
m_userAgent.ClientCallAnswered += CallAnswered;
m_userAgent.ClientCallFailed += CallFailed;
m_userAgent.OnCallHungup += CallFinished;
m_userAgent.ServerCallCancelled += IncomingCallCancelled;
m_userAgent.OnTransferNotify += OnTransferNotify;
m_userAgent.OnDtmfTone += OnDtmfTone;
}
public override void ViewDidLoad()
{
base.ViewDidLoad();
// Perform any additional setup after loading the view, typically from a nib.
Action <string> logDelegate = (str) =>
{
InvokeOnMainThread(() => {
LogTextView.Text += str;
});
};
SIPSorcery.LogFactory.Set(new TextViewLoggerFactory(logDelegate));
_userAgent = new SIPUserAgent();
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Convert Audio");
AddConsoleLogger();
//WaveFormatConversionStream converter = new WaveFormatConversionStream(_format_s16le48k, )
_waveFile = new WaveFileWriter("output_s16le48k.mp3", _format_s16le48k);
var sipTransport = new SIPTransport();
var userAgent = new SIPUserAgent(sipTransport, null);
userAgent.OnCallHungup += (dialog) =>
{
Console.WriteLine("Call hungup.");
_waveFile?.Close();
};
//EnableTraceLogs(sipTransport);
var audioOptions = new AudioSourceOptions {
AudioSource = AudioSourcesEnum.Silence
};
var audioFormats = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU
};
var rtpSession = new RtpAudioSession(audioOptions, audioFormats);
rtpSession.OnRtpPacketReceived += RtpSession_OnRtpPacketReceived;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
public async Task HandleInvalidSdpPortOnAnswerUnitTest()
{
logger.LogDebug("--> " + System.Reflection.MethodBase.GetCurrentMethod().Name);
logger.BeginScope(System.Reflection.MethodBase.GetCurrentMethod().Name);
SIPTransport transport = new SIPTransport();
SIPUserAgent userAgent = new SIPUserAgent(transport, null);
string inviteReqStr = @"INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK57441c4980b94e1686a06ae080be2935;rport
To: <sip:[email protected]>
From: <sip:0.0.0.0:0>;tag=MYILIYPHQD
Call-ID: ddf0e5a9687b4745925438da9000445d
CSeq: 1 INVITE
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE
Content-Length: 0
v=0
o=- 1838015445 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 79762 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv";
SIPEndPoint dummySipEndPoint = new SIPEndPoint(new IPEndPoint(IPAddress.Any, 0));
SIPMessageBuffer sipMessageBuffer = SIPMessageBuffer.ParseSIPMessage(inviteReqStr, dummySipEndPoint, dummySipEndPoint);
SIPRequest inviteReq = SIPRequest.ParseSIPRequest(sipMessageBuffer);
var uas = userAgent.AcceptCall(inviteReq);
RTPSession rtpSession = new RTPSession(false, false, false);
MediaStreamTrack audioTrack = new MediaStreamTrack(SDPMediaTypesEnum.audio, false, new List <SDPMediaFormat> {
new SDPMediaFormat(SDPMediaFormatsEnum.PCMU)
});
rtpSession.addTrack(audioTrack);
var result = await userAgent.Answer(uas, rtpSession);
Assert.False(result);
rtpSession.Close("normal");
}
static async Task Main()
{
Console.WriteLine("SIPSorcery Play Sounds Demo");
AddConsoleLogger();
var sipTransport = new SIPTransport();
EnableTraceLogs(sipTransport);
var userAgent = new SIPUserAgent(sipTransport, null);
var rtpSession = new WindowsAudioRtpSession();
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, rtpSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
if (callResult)
{
await Task.Delay(1000);
await rtpSession.SendAudioFromStream(new FileStream(WELCOME_16K, FileMode.Open), AudioSamplingRatesEnum.SampleRate16KHz);
await Task.Delay(1000);
await rtpSession.SendAudioFromStream(new FileStream(GOODBYE_16K, FileMode.Open), AudioSamplingRatesEnum.SampleRate16KHz);
}
Console.WriteLine("press any key to exit...");
Console.Read();
if (userAgent.IsCallActive)
{
Console.WriteLine("Hanging up.");
userAgent.Hangup();
}
// Give the hangup a chance to complete.
await Task.Delay(1000);
// Clean up.
sipTransport.Shutdown();
SIPSorcery.Net.DNSManager.Stop();
}
static async Task Main()
{
Console.WriteLine("SIP Get Started");
var userAgent = new SIPUserAgent();
var winAudio = new WindowsAudioEndPoint(new AudioEncoder());
var voipMediaSession = new VoIPMediaSession(winAudio.ToMediaEndPoints());
voipMediaSession.AcceptRtpFromAny = true;
// Place the call and wait for the result.
bool callResult = await userAgent.Call(DESTINATION, null, null, voipMediaSession);
Console.WriteLine($"Call result {((callResult) ? "success" : "failure")}.");
Console.WriteLine("Press any key to hangup and exit.");
Console.ReadLine();
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The suer agent the RTP session is being created for.</param>
/// <returns>A new RTP session object.</returns>
private static RtpAudioSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <SDPMediaFormatsEnum> codecs = new List <SDPMediaFormatsEnum> {
SDPMediaFormatsEnum.PCMU, SDPMediaFormatsEnum.PCMA, SDPMediaFormatsEnum.G722
};
var audioSource = DummyAudioSourcesEnum.SineWave;
if (!Enum.TryParse <DummyAudioSourcesEnum>(dst, out audioSource))
{
audioSource = DummyAudioSourcesEnum.Silence;
}
var audioOptions = new DummyAudioOptions {
AudioSource = audioSource
};
if (audioSource == DummyAudioSourcesEnum.Music)
{
audioOptions.SourceFiles = new Dictionary <SDPMediaFormatsEnum, string>();
if (codecs.Contains(SDPMediaFormatsEnum.PCMA))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.PCMA, MUSIC_FILE_PCMA);
}
if (codecs.Contains(SDPMediaFormatsEnum.PCMU))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.PCMU, MUSIC_FILE_PCMU);
}
if (codecs.Contains(SDPMediaFormatsEnum.G722))
{
audioOptions.SourceFiles.Add(SDPMediaFormatsEnum.G722, MUSIC_FILE_G722);
}
}
;
Log.LogInformation($"RTP audio session source set to {audioSource}.");
var rtpAudioSession = new RtpAudioSession(audioOptions, codecs);
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (type, rtp) => OnRtpPacketReceived(ua, type, rtp);
return(rtpAudioSession);
}
protected override void OnCreate(Bundle savedInstanceState)
{
base.OnCreate(savedInstanceState);
Xamarin.Essentials.Platform.Init(this, savedInstanceState);
SetContentView(Resource.Layout.activity_main);
var callButton = FindViewById <Button>(Resource.Id.callButton);
var cancelButton = FindViewById <Button>(Resource.Id.cancelButton);
var destination = FindViewById <TextView>(Resource.Id.callDestination);
var statusScroll = FindViewById <ScrollView>(Resource.Id.statusScroll);
var statusText = FindViewById <TextView>(Resource.Id.statusTextView);
Action <string> logDelegate = (str) =>
{
this.RunOnUiThread(() =>
{
statusText.Append(str);
statusScroll.FullScroll(FocusSearchDirection.Down);
});
};
SIPSorcery.LogFactory.Set(new TextViewLoggerFactory(logDelegate));
var userAgent = new SIPUserAgent();
callButton.Click += async(sender, e) =>
{
callButton.Enabled = false;
cancelButton.Enabled = true;
logDelegate($"Calling {destination.Text}...\n");
var callResult = await userAgent.Call(destination.Text, null, null, new AudioSendOnlyMediaSession());
logDelegate($"Call result {callResult}...\n");
};
cancelButton.Click += (sender, e) =>
{
cancelButton.Enabled = false;
callButton.Enabled = true;
logDelegate("Cancelled.\n");
};
}
/// <summary>
/// Because this is a server user agent the SIP transport must start listening for client user agents.
/// </summary>
private static async Task OnRequest(SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest)
{
try
{
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
SIPUserAgent ua = new SIPUserAgent(_sipTransport, null);
ua.OnCallHungup += OnHangup;
ua.ServerCallCancelled += (uas) => Log.LogDebug("Incoming call cancelled by remote party.");
ua.OnDtmfTone += (key, duration) => OnDtmfTone(ua, key, duration);
var uas = ua.AcceptCall(sipRequest);
var rtpSession = CreateRtpSession(ua);
await ua.Answer(uas, rtpSession);
if (ua.IsCallActive)
{
_calls.TryAdd(ua.Dialogue.CallId, ua);
Timer sendSilenceTimer = new Timer(SendSilence, ua, 0, SEND_SILENCE_PERIOD_MS);
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.CallLegTransactionDoesNotExist, null);
await _sipTransport.SendResponseAsync(byeResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.SUBSCRIBE)
{
SIPResponse notAllowededResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.MethodNotAllowed, null);
await _sipTransport.SendResponseAsync(notAllowededResponse);
}
else if (sipRequest.Method == SIPMethodsEnum.OPTIONS || sipRequest.Method == SIPMethodsEnum.REGISTER)
{
SIPResponse optionsResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await _sipTransport.SendResponseAsync(optionsResponse);
}
}
catch (Exception reqExcp)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
}
/// <summary>
/// Example of how to create a basic RTP session object and hook up the event handlers.
/// </summary>
/// <param name="ua">The user agent the RTP session is being created for.</param>
/// <param name="dst">THe destination specified on an incoming call. Can be used to
/// set the audio source.</param>
/// <returns>A new RTP session object.</returns>
private static VoIPMediaSession CreateRtpSession(SIPUserAgent ua, string dst)
{
List <AudioCodecsEnum> codecs = new List <AudioCodecsEnum> {
AudioCodecsEnum.PCMU, AudioCodecsEnum.PCMA, AudioCodecsEnum.G722
};
var audioSource = AudioSourcesEnum.SineWave;
if (string.IsNullOrEmpty(dst) || !Enum.IsDefined(typeof(AudioSourcesEnum), dst) || !Enum.TryParse(dst, out audioSource))
{
audioSource = AudioSourcesEnum.Music;
}
Log.LogInformation($"RTP audio session source set to {audioSource}.");
AudioExtrasSource audioExtrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions {
AudioSource = audioSource
});
audioExtrasSource.RestrictFormats(formats => codecs.Contains(formats.Codec));
var rtpAudioSession = new VoIPMediaSession(new MediaEndPoints {
AudioSource = audioExtrasSource
});
rtpAudioSession.AcceptRtpFromAny = true;
// Wire up the event handler for RTP packets received from the remote party.
rtpAudioSession.OnRtpPacketReceived += (ep, type, rtp) => OnRtpPacketReceived(ua, type, rtp);
rtpAudioSession.OnTimeout += (mediaType) =>
{
if (ua?.Dialogue != null)
{
Log.LogWarning($"RTP timeout on call with {ua.Dialogue.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
ua.Hangup();
};
return(rtpAudioSession);
}
