Closed PauloTroguilho closed 6 days ago
PauloTroguilho
commented
3 years ago Hello, I did a try catch on the method to handle the error and I'm getting new error, could you help me? thankful ! Error: Application:UserAgentServer.exe CoreCLR Version:4.7 .net core version 3.1.4 Description:The process was terminated due to an unhandled exeception. Exception Info: execption code c00000005, exception address 00007FF928F938D4
sipsorcery
commented
3 years ago Can you share your code? It seems the audio file you're attempting to play is getting closed for some reason.
PauloTroguilho
commented
3 years ago //----------------------------------------------------------------------------- // Filename: Program.cs // // Description: An example program of how to use the SIPSorcery core library to // act as the server for a SIP call. // // Author(s): // Aaron Clauson (aaron@sipsorcery.com) // // History: // 09 Oct 2019 Aaron Clauson Created, Dublin, Ireland. // 26 Feb 2020 Aaron Clauson Switched RTP to use RtpAVSession. // // License: // BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file. //-----------------------------------------------------------------------------
//----------------------------------------------------------------------------- // This example can be used with the automated SIP test tool [SIPp] (https://github.com/SIPp/sipp) // and its inbuilt User Agent Client scenario. // Note: SIPp doesn't support IPv6. // // To install on WSL: // $ sudo apt install sip-tester // // Running tests (press the '+' key while test is running to increase the call rate): // For UDP testing: sipp -sn uac 127.0.0.1 // For TCP testing: sipp -sn uac localhost -t t1 //-----------------------------------------------------------------------------
//----------------------------------------------------------------------------- // Media files: // The "Simplicity" audio used in this example is from an artist called MACROFORM // and can be downloaded directly from: https://www.jamendo.com/track/579315/simplicity?language=en // The use of the audio is licensed under the Creative Commons // https://creativecommons.org/licenses/by-nd/2.0/ // The audio is free for personal use but a license may be required for commercial use. // If it sounds familiar this particular file is also included as part of Asterisk's // (asterisk.org) music on hold. // // ffmpeg can be used to convert the mp3 file into the required format for placing directly // into the RTP packets. Currently this example supports two audio formats: G711.ULAW (or PCMU) // and G722. // // ffmpeg -i Macroform_-Simplicity.mp3 -ac 1 -ar 8k -ab 64k -f mulaw Macroform-Simplicity.ulaw // ffmpeg -i Macroform-Simplicity.mp3 -ar 16k -acodec g722 Macroform-_Simplicity.g722 //-----------------------------------------------------------------------------
using System; using System.Collections.Concurrent; using System.Collections.Generic; using System.IO; using System.Linq; using System.Net; using System.Net.Sockets; using System.Security.Cryptography.X509Certificates; using System.Threading; using System.Threading.Tasks; using Microsoft.CognitiveServices.Speech; using Microsoft.Extensions.Logging; using Microsoft.Extensions.Logging.Abstractions; using Serilog; using Serilog.Extensions.Logging; using SIPSorcery.Media; using SIPSorcery.Net; using SIPSorcery.SIP; using SIPSorcery.SIP.App; using SIPSorceryMedia.Abstractions; using SIPSorceryMedia.Windows;
namespace SIPSorcery { class Program { private static int SIP_LISTEN_PORT = 5060; private static int SIPS_LISTEN_PORT = 5060; //private static int SIP_WEBSOCKET_LISTEN_PORT = 80; //private static int SIP_SECURE_WEBSOCKET_LISTEN_PORT = 443; private static string SIPS_CERTIFICATE_PATH = "localhost.pfx"; private static string SSK = "iiiiiii"; private static string SSR = "westus"; private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance; private static ConcurrentDictionary<string, SIPServerUserAgent> _calls = new ConcurrentDictionary<string, SIPServerUserAgent>();
private static ConcurrentDictionary<string, string> _callsINVITE = new ConcurrentDictionary<string, string>();
public static FileStream dia = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream tarde = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream noite = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream gostaria = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream sevoce = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d1 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream seconhece = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d2 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream senconhece = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d3 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
static void Main(string[] args)
{
Log.LogDebug("SIPSorcery user agent server example.");
Log.LogDebug("Press h to hangup a call or ctrl-c to exit.");
Log = AddConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogDebug($"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(SIPS_CERTIFICATE_PATH);
// 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));
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Parse("10.192.8.160"), SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Parse("10.192.8.160"), SIP_LISTEN_PORT)));
//// 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.
//VoIPMediaSession 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
{
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
VoIPMediaSession rtpSession = null;
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
IEnumerable<string> list = _callsINVITE.Keys.Where(p => p == sipRequest?.Header?.CallId);
if (list.Count() > 0)
{
Log.LogDebug("List>0");
}
else
{
Log.LogDebug("Call invite");
var Name = string.Empty;
if (sipRequest?.Header?.From?.FromName == null)
{
Name = "Nulo";
}
else
{
Name = sipRequest?.Header?.From?.FromName;
}
_callsINVITE.TryAdd(sipRequest?.Header?.CallId, Name);
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
var localAudios = @"C:\URA\recoreapp_URA\";
// 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.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMA)))
{
Log.LogDebug($"Client offer contained PCMA audio codec.");
WindowsAudioEndPoint winAudio_n = new WindowsAudioEndPoint(new AudioEncoder());
// winAudio_n.RestrictCodecs(new List<AudioCodecsEnum> { AudioCodecsEnum.G722,AudioCodecsEnum.PCMA,AudioCodecsEnum.PCMU });
winAudio_n.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMA);
AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });
// rtpSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = extrasSource });
rtpSession = new VoIPMediaSession(winAudio_n.ToMediaEndPoints());
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession?.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
await sipTransport?.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
//rtpCts?.Cancel();
//rtpCts = new CancellationTokenSource();
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.Close(null);
uas?.Hangup(true);
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
};
// rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
rtpSession.OnTimeout += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSession?.Close(null);
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
};
rtpSession.OnRtcpBye += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSession?.Close(null);
// winAudio_n?.CloseAudio();
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
};
rtpSession.OnRtpClosed += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
// This app only uses each SIP user agent once so here the agent is
// explicitly closed to prevent is responding to any new SIP requests.
//uas?.Hangup(false);
rtpCts?.Cancel();
rtpSession?.Close(null);
// winAudio_n?.CloseAudio();
ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
GC.Collect();
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
if (uas.IsUASAnswered)
{
if (uas.IsHungup != true)
{
_calls.TryAdd(uas?.SIPDialogue?.CallId, uas);
}
}
try
{
var HFNS = Name.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = Name.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
Log.LogDebug(fileNamemm);
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
var config_t = SpeechConfig.FromSubscription("iiii", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
}
catch (Exception e)
{
Log.LogError(e.Message);
}
await rtpSession.Start();
await winAudio_n.PauseAudio();
// if()
//voipSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 20;
await rtpSession.AudioExtrasSource.StartAudio();
//await Task.Delay(100);
if (DateTime.Now.Hour < 12)
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(dia, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
else if (DateTime.Now.Hour < 18)
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(tarde, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
else
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(noite, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(gostaria, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
var HFN = Name;
var HFNS = HFN.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = HFN.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
//await Task.Delay(1000);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
var config_t = SpeechConfig.FromSubscription("iiiiii", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t?.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(sevoce, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d1, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(seconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(d2, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(senconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d3, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
//Task t2 = Task.Factory.StartNew(() =>
//{
// // Thread.Sleep(300);
// foreach (var item in _callsINVITE)
// {
// Log.LogDebug(item.Key + "-" + item.Value);
// // Thread.Sleep(150);
// }
//});
//try
//{
// Task.WaitAll(t2);
//}
//catch (AggregateException ex) // No exception
//{
//Log.LogWarning(ex.Flatten().Message);
//}
uas?.Hangup(false);
await rtpSession?.AudioExtrasSource.PauseAudio();
await winAudio_n?.CloseAudio();
rtpCts?.Cancel();
rtpSession?.Close(null);
//winAudio_n?.CloseAudio();
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
//// This app only uses each SIP user agent once so here the agent is
//// explicitly closed to prevent is responding to any new SIP requests.
////uas?.Hangup(false);
//rtpCts?.Cancel();
//rtpSession.Close(null);
////winAudio_n?.CloseAudio();
//ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
}
}
else if(offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
WindowsAudioEndPoint winAudio_n = new WindowsAudioEndPoint(new AudioEncoder());
// winAudio_n.RestrictCodecs(new List<AudioCodecsEnum> { AudioCodecsEnum.G722,AudioCodecsEnum.PCMA,AudioCodecsEnum.PCMU });
winAudio_n.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMU);
// AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });
// rtpSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = extrasSource });
rtpSession = new VoIPMediaSession(winAudio_n.ToMediaEndPoints());
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession?.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
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, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.Close(null);
uas?.Hangup(true);
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
};
// rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
rtpSession.OnTimeout += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSession?.Close(null);
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
};
rtpSession.OnRtcpBye += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSession?.Close(null);
// winAudio_n?.CloseAudio();
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
};
rtpSession.OnRtpClosed += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
// This app only uses each SIP user agent once so here the agent is
// explicitly closed to prevent is responding to any new SIP requests.
//uas?.Hangup(false);
rtpCts?.Cancel();
rtpSession?.Close(null);
// winAudio_n?.CloseAudio();
ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
GC.Collect();
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
if (uas.IsUASAnswered)
{
if (uas.IsHungup != true)
{
_calls.TryAdd(uas?.SIPDialogue?.CallId, uas);
}
}
try
{
var HFNS = Name.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = Name.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
Log.LogDebug(fileNamemm);
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
var config_t = SpeechConfig.FromSubscription("iiiiiiii", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
}
catch (Exception e)
{
Log.LogError(e.Message);
}
await rtpSession.Start();
await winAudio_n.PauseAudio();
// if()
//voipSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 20;
await rtpSession.AudioExtrasSource.StartAudio();
//await Task.Delay(100);
if (DateTime.Now.Hour < 12)
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(dia, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
else if (DateTime.Now.Hour < 18)
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(tarde, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
else
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(noite, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(gostaria, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
var HFN = Name;
var HFNS = HFN.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = HFN.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
//await Task.Delay(1000);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
var config_t = SpeechConfig.FromSubscription("iiiiii", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t?.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(sevoce, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d1, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(seconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(d2, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(senconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d3, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
//Task t2 = Task.Factory.StartNew(() =>
//{
// // Thread.Sleep(300);
// foreach (var item in _callsINVITE)
// {
// Log.LogDebug(item.Key + "-" + item.Value);
// // Thread.Sleep(150);
// }
//});
//try
//{
// Task.WaitAll(t2);
//}
//catch (AggregateException ex) // No exception
//{
//Log.LogWarning(ex.Flatten().Message);
//}
uas?.Hangup(false);
await rtpSession?.AudioExtrasSource.PauseAudio();
await winAudio_n?.CloseAudio();
rtpCts?.Cancel();
rtpSession?.Close(null);
//winAudio_n?.CloseAudio();
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
//// This app only uses each SIP user agent once so here the agent is
//// explicitly closed to prevent is responding to any new SIP requests.
////uas?.Hangup(false);
//rtpCts?.Cancel();
//rtpSession.Close(null);
////winAudio_n?.CloseAudio();
//ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
}
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Log.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.Close(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)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
//Hangup(uas).Wait();
//rtpSession?.Close(null);
//rtpCts?.Cancel();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
//Hangup(uas).Wait();
//rtpSession?.Close(null);
//rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
Log.LogInformation("Quitting...");
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
/// <summary>
/// Hangs up the current call.
/// </summary>
/// <param name="uas">The user agent server to hangup the call on.</param>
private static async Task Hangup(SIPServerUserAgent uas)
{
try
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to hangup...");
await Task.Delay(1000);
}
}
catch (Exception excp)
{
Log.LogError($"Exception Hangup. {excp.Message}");
}
}
/// <summary>
/// Enable detailed SIP log messages.
/// </summary>
private static void EnableTraceLogs(SIPTransport sipTransport)
{
sipTransport.SIPRequestInTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request received: {localEP}<-{remoteEP}");
Log.LogDebug(req.ToString());
};
sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request sent: {localEP}->{remoteEP}");
Log.LogDebug(req.ToString());
};
sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response received: {localEP}<-{remoteEP}");
Log.LogDebug(resp.ToString());
};
sipTransport.SIPResponseOutTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response sent: {localEP}->{remoteEP}");
Log.LogDebug(resp.ToString());
};
sipTransport.SIPRequestRetransmitTraceEvent += (tx, req, count) =>
{
Log.LogDebug($"Request retransmit {count} for request {req.StatusLine}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
sipTransport.SIPResponseRetransmitTraceEvent += (tx, resp, count) =>
{
Log.LogDebug($"Response retransmit {count} for response {resp.ShortDescription}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
}
/// <summary>
/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
/// </summary>
private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
{
var serilogLogger = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
var factory = new SerilogLoggerFactory(serilogLogger);
SIPSorcery.LogFactory.Set(factory);
return factory.CreateLogger<Program>();
}
}}
PauloTroguilho
commented
3 years ago of course, follow the code above, thanks for the feedback! if you can evaluate and inform if it would hold up to 1000 assistance, I would be grateful
PauloTroguilho
commented
3 years ago explaining the flow, the customer's name is vocalized in the service, if the file does not exist it generates a file with azure, the process of generating this with more than 2000 files in a folder.
PauloTroguilho
commented
3 years ago add try cath in stopsendfromaudiostrem by error possible https://github.com/sipsorcery-org/sipsorcery/issues/550 , I try
private void StopSendFromAudioStream() { try { if (_streamSendInProgress) { lock (_streamSourceTimer) { _streamSourceTimer?.Dispose(); _streamSendInProgress = false;
OnSendFromAudioStreamComplete?.Invoke();
}
}
}
catch (Exception e)
{
Console.Write(e.Message);
}
}Wow that's a lot of code! I'm pretty sure ou could simplify that significantly.
The first thing I'd recommend is only checking uas?.IsUASAnswered once. After that check uas?.IsHungup. You need to check that the call hasn't been hungup by the remote party each time you start playing a new audio file.
PauloTroguilho
commented
3 years ago //----------------------------------------------------------------------------- // Filename: Program.cs // // Description: An example program of how to use the SIPSorcery core library to // act as the server for a SIP call. // // Author(s): // Aaron Clauson (aaron@sipsorcery.com) // // History: // 09 Oct 2019 Aaron Clauson Created, Dublin, Ireland. // 26 Feb 2020 Aaron Clauson Switched RTP to use RtpAVSession. // // License: // BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file. //-----------------------------------------------------------------------------
//----------------------------------------------------------------------------- // This example can be used with the automated SIP test tool [SIPp] (https://github.com/SIPp/sipp) // and its inbuilt User Agent Client scenario. // Note: SIPp doesn't support IPv6. // // To install on WSL: // $ sudo apt install sip-tester // // Running tests (press the '+' key while test is running to increase the call rate): // For UDP testing: sipp -sn uac 127.0.0.1 // For TCP testing: sipp -sn uac localhost -t t1 //-----------------------------------------------------------------------------
//----------------------------------------------------------------------------- // Media files: // The "Simplicity" audio used in this example is from an artist called MACROFORM // and can be downloaded directly from: https://www.jamendo.com/track/579315/simplicity?language=en // The use of the audio is licensed under the Creative Commons // https://creativecommons.org/licenses/by-nd/2.0/ // The audio is free for personal use but a license may be required for commercial use. // If it sounds familiar this particular file is also included as part of Asterisk's // (asterisk.org) music on hold. // // ffmpeg can be used to convert the mp3 file into the required format for placing directly // into the RTP packets. Currently this example supports two audio formats: G711.ULAW (or PCMU) // and G722. // // ffmpeg -i Macroform_-Simplicity.mp3 -ac 1 -ar 8k -ab 64k -f mulaw Macroform-Simplicity.ulaw // ffmpeg -i Macroform-Simplicity.mp3 -ar 16k -acodec g722 Macroform-_Simplicity.g722 //-----------------------------------------------------------------------------
using System; using System.Collections.Concurrent; using System.Collections.Generic; using System.IO; using System.Linq; using System.Net; using System.Net.Sockets; using System.Security.Cryptography.X509Certificates; using System.Threading; using System.Threading.Tasks; using Microsoft.CognitiveServices.Speech; using Microsoft.Extensions.Configuration; using Microsoft.Extensions.Logging; using Microsoft.Extensions.Logging.Abstractions; using Serilog; using Serilog.Extensions.Logging; using SIPSorcery.Media; using SIPSorcery.Net; using SIPSorcery.SIP; using SIPSorcery.SIP.App; using SIPSorceryMedia.Abstractions; using SIPSorceryMedia.Windows;
namespace SIPSorcery { class Program { private static int SIP_LISTEN_PORT = 5060; private static int SIPS_LISTEN_PORT = 5060; //private static int SIP_WEBSOCKET_LISTEN_PORT = 80; //private static int SIP_SECURE_WEBSOCKET_LISTEN_PORT = 443; private static string SIPS_CERTIFICATE_PATH = "localhost.pfx"; private static string SSK = "oooo"; private static string SSR = "westus"; private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance; private static ConcurrentDictionary<string, SIPServerUserAgent> _calls = new ConcurrentDictionary<string, SIPServerUserAgent>();
private static ConcurrentDictionary<string, string> _callsINVITE = new ConcurrentDictionary<string, string>();
public static FileStream dia = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream tarde = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream noite = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream gostaria = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream sevoce = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d1 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream seconhece = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d2 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream senconhece = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d3 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
static void Main(string[] args)
{
Log.LogDebug("SIPSorcery user agent server example.");
Log.LogDebug("Press h to hangup a call or ctrl-c to exit.");
Log = AddConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogDebug($"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(SIPS_CERTIFICATE_PATH);
// 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));
///
var builder = new ConfigurationBuilder()
.SetBasePath(Directory.GetCurrentDirectory())
.AddJsonFile("appsettings.json");
var configuration = builder.Build();
//string myKey1 = configuration["myKey1"];
//Console.WriteLine(myKey1);
string ip = configuration.GetSection("ip").Value;
Console.WriteLine(ip);
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Parse(ip), SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Parse(ip), SIP_LISTEN_PORT)));
//// 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.
//VoIPMediaSession 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
{
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
VoIPMediaSession rtpSession = null;
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
IEnumerable<string> list = _callsINVITE.Keys.Where(p => p == sipRequest?.Header?.CallId);
if (list.Count() > 0)
{
Log.LogDebug("List>0");
}
else
{
Log.LogDebug("Call invite");
var Name = string.Empty;
if (sipRequest?.Header?.From?.FromName == null)
{
Name = "Nulo";
}
else
{
Name = sipRequest?.Header?.From?.FromName;
}
_callsINVITE.TryAdd(sipRequest?.Header?.CallId, Name);
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
var localAudios = @"C:\URA\recoreapp_URA\";
// 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.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMA)))
{
Log.LogDebug($"Client offer contained PCMA audio codec.");
WindowsAudioEndPoint winAudio_n = new WindowsAudioEndPoint(new AudioEncoder());
// winAudio_n.RestrictCodecs(new List<AudioCodecsEnum> { AudioCodecsEnum.G722,AudioCodecsEnum.PCMA,AudioCodecsEnum.PCMU });
winAudio_n.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMA);
AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });
// rtpSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = extrasSource });
rtpSession = new VoIPMediaSession(winAudio_n.ToMediaEndPoints());
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession?.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
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, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
//rtpCts?.Cancel();
//rtpSession.Close(null);
//uas?.Hangup(true);
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
}
};
// rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
rtpSession.OnTimeout += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
}
};
rtpSession.OnRtcpBye += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
//uas?.Hangup(true);
//rtpCts?.Cancel();
//rtpSession?.Close(null);
// winAudio_n?.CloseAudio();
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
};
rtpSession.OnRtpClosed += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
// This app only uses each SIP user agent once so here the agent is
// explicitly closed to prevent is responding to any new SIP requests.
//uas?.Hangup(false);
// rtpCts?.Cancel();
// rtpSession?.Close(null);
//// winAudio_n?.CloseAudio();
// ua?.Hangup(true);
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
GC.Collect();
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
if (uas.IsUASAnswered)
{
if (uas.IsHungup != true)
{
_calls.TryAdd(uas?.SIPDialogue?.CallId, uas);
}
}
try
{
var HFNS = Name.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = Name.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
Log.LogDebug(fileNamemm);
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
var config_t = SpeechConfig.FromSubscription("000", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
}
catch (Exception e)
{
Log.LogError(e.Message);
}
await rtpSession.Start();
await winAudio_n.PauseAudio();
// if()
//voipSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 20;
await rtpSession.AudioExtrasSource.StartAudio();
//await Task.Delay(100);
if (DateTime.Now.Hour < 12)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(dia, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
else if (DateTime.Now.Hour < 18)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(tarde, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
else
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(noite, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(gostaria, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
var HFN = Name;
var HFNS = HFN.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = HFN.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
//await Task.Delay(1000);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
var config_t = SpeechConfig.FromSubscription("00", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t?.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(sevoce, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d1, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(seconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(d2, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(senconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d3, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//Task t2 = Task.Factory.StartNew(() =>
//{
// // Thread.Sleep(300);
// foreach (var item in _callsINVITE)
// {
// Log.LogDebug(item.Key + "-" + item.Value);
// // Thread.Sleep(150);
// }
//});
//try
//{
// Task.WaitAll(t2);
//}
//catch (AggregateException ex) // No exception
//{
//Log.LogWarning(ex.Flatten().Message);
//}
uas?.Hangup(false);
await rtpSession?.AudioExtrasSource.PauseAudio();
await winAudio_n?.CloseAudio();
rtpCts?.Cancel();
rtpSession?.Close(null);
//winAudio_n?.CloseAudio();
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
//// This app only uses each SIP user agent once so here the agent is
//// explicitly closed to prevent is responding to any new SIP requests.
////uas?.Hangup(false);
//rtpCts?.Cancel();
//rtpSession.Close(null);
////winAudio_n?.CloseAudio();
//ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
}
}
else if(offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMU audio codec.");
WindowsAudioEndPoint winAudio_n = new WindowsAudioEndPoint(new AudioEncoder());
// winAudio_n.RestrictCodecs(new List<AudioCodecsEnum> { AudioCodecsEnum.G722,AudioCodecsEnum.PCMA,AudioCodecsEnum.PCMU });
winAudio_n.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMU);
// AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });
// rtpSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = extrasSource });
rtpSession = new VoIPMediaSession(winAudio_n.ToMediaEndPoints());
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession?.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
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, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
rtpCts?.Cancel();
rtpSession.Close(null);
uas?.Hangup(true);
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
};
// rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
rtpSession.OnTimeout += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSession?.Close(null);
}
}
}
};
rtpSession.OnRtcpBye += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
// winAudio_n?.CloseAudio();
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
uas?.Hangup(true);
rtpCts?.Cancel();
rtpSession?.Close(null);
}
}
};
rtpSession.OnRtpClosed += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
// This app only uses each SIP user agent once so here the agent is
// explicitly closed to prevent is responding to any new SIP requests.
//uas?.Hangup(false);
rtpCts?.Cancel();
rtpSession?.Close(null);
// winAudio_n?.CloseAudio();
ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
GC.Collect();
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
if (uas.IsUASAnswered)
{
if (uas.IsHungup != true)
{
_calls.TryAdd(uas?.SIPDialogue?.CallId, uas);
}
}
try
{
var HFNS = Name.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = Name.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
Log.LogDebug(fileNamemm);
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
var config_t = SpeechConfig.FromSubscription("00", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
}
catch (Exception e)
{
Log.LogError(e.Message);
}
await rtpSession.Start();
await winAudio_n.PauseAudio();
// if()
//voipSession.AudioExtrasSource.AudioSamplePeriodMilliseconds = 20;
await rtpSession.AudioExtrasSource.StartAudio();
//await Task.Delay(100);
if (DateTime.Now.Hour < 12)
{
if (uas?.IsUASAnswered == true)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(dia, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
}
else if (DateTime.Now.Hour < 18)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(tarde, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
else
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(noite, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(gostaria, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
var HFN = Name;
var HFNS = HFN.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = HFN.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
if (!File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
//await Task.Delay(1000);
// logger.Debug("synsthesizer");
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
var config_t = SpeechConfig.FromSubscription("00", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t?.SaveToWaveFileAsync(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(sevoce, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d1, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(seconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(d2, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(senconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d3, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//Task t2 = Task.Factory.StartNew(() =>
//{
// // Thread.Sleep(300);
// foreach (var item in _callsINVITE)
// {
// Log.LogDebug(item.Key + "-" + item.Value);
// // Thread.Sleep(150);
// }
//});
//try
//{
// Task.WaitAll(t2);
//}
//catch (AggregateException ex) // No exception
//{
//Log.LogWarning(ex.Flatten().Message);
//}
uas?.Hangup(false);
await rtpSession?.AudioExtrasSource.PauseAudio();
await winAudio_n?.CloseAudio();
rtpCts?.Cancel();
rtpSession?.Close(null);
//winAudio_n?.CloseAudio();
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
//// This app only uses each SIP user agent once so here the agent is
//// explicitly closed to prevent is responding to any new SIP requests.
////uas?.Hangup(false);
//rtpCts?.Cancel();
//rtpSession.Close(null);
////winAudio_n?.CloseAudio();
//ua?.Hangup(true);
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
}
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Log.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.Close(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)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
//Hangup(uas).Wait();
//rtpSession?.Close(null);
//rtpCts?.Cancel();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
//Hangup(uas).Wait();
//rtpSession?.Close(null);
//rtpCts?.Cancel();
}
if (keyProps.KeyChar == 'q')
{
Log.LogInformation("Quitting...");
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
/// <summary>
/// Hangs up the current call.
/// </summary>
/// <param name="uas">The user agent server to hangup the call on.</param>
private static async Task Hangup(SIPServerUserAgent uas)
{
try
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to hangup...");
await Task.Delay(1000);
}
}
catch (Exception excp)
{
Log.LogError($"Exception Hangup. {excp.Message}");
}
}
/// <summary>
/// Enable detailed SIP log messages.
/// </summary>
private static void EnableTraceLogs(SIPTransport sipTransport)
{
sipTransport.SIPRequestInTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request received: {localEP}<-{remoteEP}");
Log.LogDebug(req.ToString());
};
sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request sent: {localEP}->{remoteEP}");
Log.LogDebug(req.ToString());
};
sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response received: {localEP}<-{remoteEP}");
Log.LogDebug(resp.ToString());
};
sipTransport.SIPResponseOutTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response sent: {localEP}->{remoteEP}");
Log.LogDebug(resp.ToString());
};
sipTransport.SIPRequestRetransmitTraceEvent += (tx, req, count) =>
{
Log.LogDebug($"Request retransmit {count} for request {req.StatusLine}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
sipTransport.SIPResponseRetransmitTraceEvent += (tx, resp, count) =>
{
Log.LogDebug($"Response retransmit {count} for response {resp.ShortDescription}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
}
/// <summary>
/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
/// </summary>
private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
{
var serilogLogger = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
var factory = new SerilogLoggerFactory(serilogLogger);
SIPSorcery.LogFactory.Set(factory);
return factory.CreateLogger<Program>();
}
}}
PauloTroguilho
commented
3 years ago Of course, all guidance is welcome, I made the adjustments.... checking uas?.IsUASAnswered
PauloTroguilho
commented
3 years ago error persist, 6 to 11 times a day, the bad thing is that he closes the application :/ Error: Application:UserAgentServer.exe CoreCLR Version:4.7 .net core version 3.1.4 Description:The process was terminated due to an unhandled exeception. Exception Info: execption code c00000005, exception address 00007FF928F938D4
sipsorcery
commented
3 years ago Your code is too big for me to try and identify the problem. If you can provide a smaller sample with the exception maybe I could help. It does seem like the issue is being caused by a file being accessed after it's been closed. Maybe try putting some locks around your file access calls.
PauloTroguilho
commented
3 years ago could you give me an example please?
I will decrease the code and test blocking
sipsorcery
commented
3 years ago The AudioExtrasSource class uses locks to control access to critical resources.
PauloTroguilho
commented
3 years ago hello I did the lock but it stopped playing, I'm providing a code if you can help me I would like to understand and solve this error to stop closing automatically
Grateful
code
//----------------------------------------------------------------------------- // Filename: Program.cs // // Description: An example program of how to use the SIPSorcery core library to // act as the server for a SIP call. // // Author(s): // Aaron Clauson (aaron@sipsorcery.com) // // History: // 09 Oct 2019 Aaron Clauson Created, Dublin, Ireland. // 26 Feb 2020 Aaron Clauson Switched RTP to use RtpAVSession. // // License: // BSD 3-Clause "New" or "Revised" License, see included LICENSE.md file. //-----------------------------------------------------------------------------
//----------------------------------------------------------------------------- // This example can be used with the automated SIP test tool [SIPp] (https://github.com/SIPp/sipp) // and its inbuilt User Agent Client scenario. // Note: SIPp doesn't support IPv6. // // To install on WSL: // $ sudo apt install sip-tester // // Running tests (press the '+' key while test is running to increase the call rate): // For UDP testing: sipp -sn uac 127.0.0.1 // For TCP testing: sipp -sn uac localhost -t t1 //-----------------------------------------------------------------------------
//----------------------------------------------------------------------------- // Media files: // The "Simplicity" audio used in this example is from an artist called MACROFORM // and can be downloaded directly from: https://www.jamendo.com/track/579315/simplicity?language=en // The use of the audio is licensed under the Creative Commons // https://creativecommons.org/licenses/by-nd/2.0/ // The audio is free for personal use but a license may be required for commercial use. // If it sounds familiar this particular file is also included as part of Asterisk's // (asterisk.org) music on hold. // // ffmpeg can be used to convert the mp3 file into the required format for placing directly // into the RTP packets. Currently this example supports two audio formats: G711.ULAW (or PCMU) // and G722. // // ffmpeg -i Macroform_-Simplicity.mp3 -ac 1 -ar 8k -ab 64k -f mulaw Macroform-Simplicity.ulaw // ffmpeg -i Macroform-Simplicity.mp3 -ar 16k -acodec g722 Macroform-_Simplicity.g722 //-----------------------------------------------------------------------------
using System; using System.Collections.Concurrent; using System.Collections.Generic; using System.IO; using System.Linq; using System.Net; using System.Net.Sockets; using System.Security.Cryptography.X509Certificates; using System.Threading; using System.Threading.Tasks; using Microsoft.CognitiveServices.Speech; using Microsoft.Extensions.Configuration; using Microsoft.Extensions.Logging; using Microsoft.Extensions.Logging.Abstractions; using Serilog; using Serilog.Extensions.Logging; using SIPSorcery.Media; using SIPSorcery.Net; using SIPSorcery.SIP; using SIPSorcery.SIP.App; using SIPSorceryMedia.Abstractions; using SIPSorceryMedia.Windows;
namespace SIPSorcery { class Program { private static int SIP_LISTEN_PORT = 5060; private static int SIPS_LISTEN_PORT = 5060; private static string SIPS_CERTIFICATE_PATH = "localhost.pfx"; private static string SSK = ""; private static string SSR = "westus"; private static Microsoft.Extensions.Logging.ILogger Log = NullLogger.Instance; private static ConcurrentDictionary<string, SIPServerUserAgent> _calls = new ConcurrentDictionary<string, SIPServerUserAgent>();
private static ConcurrentDictionary<string, string> _callsINVITE = new ConcurrentDictionary<string, string>();
public static FileStream dia = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream tarde = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream noite = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream gostaria = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream sevoce = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d1 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream seconhece = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d2 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream senconhece = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
public static FileStream d3 = new FileStream(@"C:\URA\recoreapp_URA\" + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read);
static void Main(string[] args)
{
Log.LogDebug("SIPSorcery user agent server example.");
Log.LogDebug("Press h to hangup a call or ctrl-c to exit.");
Log = AddConsoleLogger();
IPAddress listenAddress = IPAddress.Any;
IPAddress listenIPv6Address = IPAddress.IPv6Any;
if (args != null && args.Length > 0)
{
if (!IPAddress.TryParse(args[0], out var customListenAddress))
{
Log.LogDebug($"Command line argument could not be parsed as an IP address \"{args[0]}\"");
listenAddress = IPAddress.Any;
}
else
{
if (customListenAddress.AddressFamily == AddressFamily.InterNetwork)
{
listenAddress = customListenAddress;
}
if (customListenAddress.AddressFamily == AddressFamily.InterNetworkV6)
{
listenIPv6Address = customListenAddress;
}
}
}
// Set up a default SIP transport.
var sipTransport = new SIPTransport();
///
var builder = new ConfigurationBuilder()
.SetBasePath(Directory.GetCurrentDirectory())
.AddJsonFile("appsettings.json");
var configuration = builder.Build();
string ip = configuration.GetSection("ip").Value;
Console.WriteLine(ip);
sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Parse(ip), SIP_LISTEN_PORT)));
sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Parse(ip), SIP_LISTEN_PORT)));
//// IPv6 channels.
EnableTraceLogs(sipTransport);
string executableDir = Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().Location);
// To keep things a bit simpler this example only supports a single call at a time and the SIP server user agent
// acts as a singleton
// Because this is a server user agent the SIP transport must start listening for client user agents.
sipTransport.SIPTransportRequestReceived += async (SIPEndPoint localSIPEndPoint, SIPEndPoint remoteEndPoint, SIPRequest sipRequest) =>
{
try
{
SIPServerUserAgent uas = null;
CancellationTokenSource rtpCts = null; // Cancellation token to stop the RTP stream.
VoIPMediaSession rtpSession = null;
if (sipRequest.Method == SIPMethodsEnum.INVITE)
{
IEnumerable<string> list = _callsINVITE.Keys.Where(p => p == sipRequest?.Header?.CallId);
if (list.Count() > 0)
{
Log.LogDebug("List>0");
}
else
{
Log.LogDebug("Call invite");
var Name = string.Empty;
if (sipRequest?.Header?.From?.FromName == null)
{
Name = "Nulo";
}
else
{
Name = sipRequest?.Header?.From?.FromName;
}
_callsINVITE.TryAdd(sipRequest?.Header?.CallId, Name);
Log.LogInformation($"Incoming call request: {localSIPEndPoint}<-{remoteEndPoint} {sipRequest.URI}.");
var localAudios = @"C:\URA\recoreapp_URA\";
// 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.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMA)) || offerSdp.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaFormats.Any(x => x.Key == (int)SDPWellKnownMediaFormatsEnum.PCMU)))
{
Log.LogDebug($"Client offer contained PCMA audio codec.");
WindowsAudioEndPoint winAudio_n = new WindowsAudioEndPoint(new AudioEncoder());
winAudio_n.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMA || x.Codec == AudioCodecsEnum.PCMU);
// winAudio_n.RestrictFormats(x => x.Codec == AudioCodecsEnum.PCMA);
AudioExtrasSource extrasSource = new AudioExtrasSource(new AudioEncoder(), new AudioSourceOptions { AudioSource = AudioSourcesEnum.Music });
// rtpSession = new VoIPMediaSession(new MediaEndPoints { AudioSource = extrasSource });
rtpSession = new VoIPMediaSession(winAudio_n.ToMediaEndPoints());
rtpSession.AcceptRtpFromAny = true;
var setResult = rtpSession?.SetRemoteDescription(SdpType.offer, offerSdp);
if (setResult != SetDescriptionResultEnum.OK)
{
// Didn't get a match on the codecs we support.
SIPResponse noMatchingCodecResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.NotAcceptableHere, setResult.ToString());
await sipTransport?.SendResponseAsync(noMatchingCodecResponse);
}
else
{
// If there's already a call in progress hang it up. Of course this is not ideal for a real softphone or server but it
// means this example can be kept simpler.
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
}
UASInviteTransaction uasTransaction = new UASInviteTransaction(sipTransport, sipRequest, null);
uas = new SIPServerUserAgent(sipTransport, null, uasTransaction, null);
uas.CallCancelled += (uasAgent) =>
{
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
}
};
// rtpSession.OnRtpClosed += (reason) => uas?.Hangup(false);
rtpSession.OnTimeout += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
}
};
rtpSession.OnRtcpBye += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
string callID = uas?.SIPDialogue?.CallId;
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
};
rtpSession.OnRtpClosed += (mediaType) =>
{
if (uas?.SIPDialogue != null)
{
Log.LogWarning($"RTP timeout on call with {uas?.SIPDialogue?.RemoteTarget}, hanging up.");
}
else
{
Log.LogWarning($"RTP timeout on incomplete call, closing RTP session.");
}
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
// This app only uses each SIP user agent once so here the agent is
// explicitly closed to prevent is responding to any new SIP requests.
if (uas != null)
{
uas?.Hangup(true);
}
if (rtpCts != null)
{
rtpCts?.Cancel();
}
if (rtpSession != null)
{
rtpSession?.Close(null);
}
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
GC.Collect();
};
uas.Progress(SIPResponseStatusCodesEnum.Trying, null, null, null, null);
// await Task.Delay(100);
uas.Progress(SIPResponseStatusCodesEnum.Ringing, null, null, null, null);
// await Task.Delay(100);
var answerSdp = rtpSession.CreateAnswer(null);
uas.Answer(SDP.SDP_MIME_CONTENTTYPE, answerSdp.ToString(), null, SIPDialogueTransferModesEnum.NotAllowed);
if (uas.IsUASAnswered)
{
if (uas.IsHungup != true)
{
_calls.TryAdd(uas?.SIPDialogue?.CallId, uas);
}
}
try
{
var HFNS1 = Name.Replace(" ", "_").Split("_");
var HFNSN1 = string.Empty;
if (HFNS1.Count() > 1)
{
HFNSN1 = HFNS1[0];
}
else
{
HFNSN1 = Name.Replace(" ", "_");
}
string fileNamemm1 = HFNSN1.Replace(" ", "_") + ".wav";
Log.LogDebug(fileNamemm1);
if (!File.Exists(localAudios + "\\" + HFNSN1.Replace(" ", "_") + ".wav"))
{
var config_t = SpeechConfig.FromSubscription("", "westus");
config_t.SpeechRecognitionLanguage = "pt-BR";
config_t.SpeechSynthesisLanguage = "pt-BR";
config_t.SetSpeechSynthesisOutputFormat(SpeechSynthesisOutputFormat.Raw8Khz16BitMonoPcm);
// using var synthesizer_t = new SpeechSynthesizer(config_t, null);
using (var synthesizer_t = new SpeechSynthesizer(config_t, null))
{
Log.LogDebug("Synt Inicio");
// logger.Debug("resultt");
var result__t = await synthesizer_t.SpeakTextAsync(HFNSN1);
if (result__t.Reason == ResultReason.SynthesizingAudioCompleted)
{
Log.LogDebug("Complete");
// using var stream = AudioDataStream.FromResult(result);
using (var audioDataStream_t = AudioDataStream.FromResult(result__t))
{
// You can save all the data in the audio data stream to a file
Log?.LogDebug("Save");
await audioDataStream_t.SaveToWaveFileAsync(localAudios + "\\" + HFNSN1.Replace(" ", "_") + ".wav");
}
}
else if (result__t.Reason == ResultReason.Canceled)
{
var cancellation = SpeechSynthesisCancellationDetails.FromResult(result__t);
Log.LogDebug($"CANCELED: Reason={cancellation.Reason}");
if (cancellation.Reason == CancellationReason.Error)
{
Log.LogDebug($"CANCELED: ErrorCode={cancellation.ErrorCode}");
Log.LogDebug($"CANCELED: ErrorDetails=[{cancellation.ErrorDetails}]");
}
}
}
}
}
catch (Exception e)
{
Log.LogError(e.Message);
}
await rtpSession.Start();
await winAudio_n.PauseAudio();
await rtpSession.AudioExtrasSource.StartAudio();
//await Task.Delay(100);
if (DateTime.Now.Hour < 12)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(dia, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "dia.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
else if (DateTime.Now.Hour < 18)
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(tarde, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "tarde.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
else
{
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(noite, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "noite.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
await rtpSession?.AudioExtrasSource.SendAudioFromStream(gostaria, AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "gostaria.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
var HFN = Name;
var HFNS = HFN.Replace(" ", "_").Split("_");
var HFNSN = string.Empty;
if (HFNS.Count() > 1)
{
HFNSN = HFNS[0];
}
else
{
HFNSN = HFN.Replace(" ", "_");
}
string fileNamemm = HFNSN.Replace(" ", "_") + ".wav";
if (File.Exists(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav"))
{
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + HFNSN.Replace(" ", "_") + ".wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
}
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(sevoce, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "sevoce.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d1, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d1.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(seconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "seconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(d2, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d2.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
// await rtpSession?.AudioExtrasSource.SendAudioFromStream(senconhece, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "senconhece.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
//await rtpSession?.AudioExtrasSource.SendAudioFromStream(d3, AudioSamplingRatesEnum.Rate8KHz);
await rtpSession?.AudioExtrasSource.SendAudioFromStream(new FileStream(localAudios + "\\" + "d3.wav", FileMode.Open, FileAccess.Read, FileShare.Read), AudioSamplingRatesEnum.Rate8KHz);
uas?.Hangup(false);
await rtpSession?.AudioExtrasSource.PauseAudio();
await winAudio_n?.CloseAudio();
rtpCts?.Cancel();
rtpSession?.Close(null);
//winAudio_n?.CloseAudio();
if (uas?.SIPDialogue != null)
{
string callID = uas?.SIPDialogue?.CallId;
if (_calls.ContainsKey(callID))
{
if (_calls.TryRemove(callID, out var ua))
{
}
}
if (_callsINVITE.ContainsKey(callID))
{
if (_callsINVITE.TryRemove(callID, out var ua))
{
}
}
}
}
}
}
}
else if (sipRequest.Method == SIPMethodsEnum.BYE)
{
Log.LogInformation("Call hungup.");
SIPResponse byeResponse = SIPResponse.GetResponse(sipRequest, SIPResponseStatusCodesEnum.Ok, null);
await sipTransport.SendResponseAsync(byeResponse);
uas?.Hangup(true);
rtpSession?.Close(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)
{
Log.LogWarning($"Exception handling {sipRequest.Method}. {reqExcp.Message}");
}
};
ManualResetEvent exitMre = new ManualResetEvent(false);
Console.CancelKeyPress += delegate (object sender, ConsoleCancelEventArgs e)
{
e.Cancel = true;
Log.LogInformation("Exiting...");
//Hangup(uas).Wait();
//rtpSession?.Close(null);
//rtpCts?.Cancel();
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
};
// Task to handle user key presses.
Task.Run(() =>
{
try
{
while (!exitMre.WaitOne(0))
{
var keyProps = Console.ReadKey();
if (keyProps.KeyChar == 'h' || keyProps.KeyChar == 'q')
{
Console.WriteLine();
Console.WriteLine("Hangup requested by user...");
}
if (keyProps.KeyChar == 'q')
{
Log.LogInformation("Quitting...");
if (sipTransport != null)
{
Log.LogInformation("Shutting down SIP transport...");
sipTransport.Shutdown();
}
exitMre.Set();
}
}
}
catch (Exception excp)
{
Log.LogError($"Exception Key Press listener. {excp.Message}.");
}
});
exitMre.WaitOne();
}
/// <summary>
/// Hangs up the current call.
/// </summary>
/// <param name="uas">The user agent server to hangup the call on.</param>
private static async Task Hangup(SIPServerUserAgent uas)
{
try
{
if (uas?.IsHungup == false)
{
uas?.Hangup(false);
// Give the BYE or CANCEL request time to be transmitted.
Log.LogInformation("Waiting 1s for call to hangup...");
await Task.Delay(1000);
}
}
catch (Exception excp)
{
Log.LogError($"Exception Hangup. {excp.Message}");
}
}
/// <summary>
/// Enable detailed SIP log messages.
/// </summary>
private static void EnableTraceLogs(SIPTransport sipTransport)
{
sipTransport.SIPRequestInTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request received: {localEP}<-{remoteEP}");
Log.LogDebug(req.ToString());
};
sipTransport.SIPRequestOutTraceEvent += (localEP, remoteEP, req) =>
{
Log.LogDebug($"Request sent: {localEP}->{remoteEP}");
Log.LogDebug(req.ToString());
};
sipTransport.SIPResponseInTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response received: {localEP}<-{remoteEP}");
Log.LogDebug(resp.ToString());
};
sipTransport.SIPResponseOutTraceEvent += (localEP, remoteEP, resp) =>
{
Log.LogDebug($"Response sent: {localEP}->{remoteEP}");
Log.LogDebug(resp.ToString());
};
sipTransport.SIPRequestRetransmitTraceEvent += (tx, req, count) =>
{
Log.LogDebug($"Request retransmit {count} for request {req.StatusLine}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
sipTransport.SIPResponseRetransmitTraceEvent += (tx, resp, count) =>
{
Log.LogDebug($"Response retransmit {count} for response {resp.ShortDescription}, initial transmit {DateTime.Now.Subtract(tx.InitialTransmit).TotalSeconds.ToString("0.###")}s ago.");
};
}
/// <summary>
/// Adds a console logger. Can be omitted if internal SIPSorcery debug and warning messages are not required.
/// </summary>
private static Microsoft.Extensions.Logging.ILogger AddConsoleLogger()
{
var serilogLogger = new LoggerConfiguration()
.Enrich.FromLogContext()
.MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
.WriteTo.Console()
.CreateLogger();
var factory = new SerilogLoggerFactory(serilogLogger);
SIPSorcery.LogFactory.Set(factory);
return factory.CreateLogger<Program>();
}
}}
PauloTroguilho
commented
3 years ago Lock
public Task SendAudioFromStream(Stream audioStream, AudioSamplingRatesEnum streamSampleRate) { if (!_isClosed && audioStream != null && audioStream.Length > 0) { lock (_sendSampleTimer) { // Stop any existing send from stream operation. StopSendFromAudioStream();
TaskCompletionSource<bool> tcs = new TaskCompletionSource<bool>(TaskCreationOptions.RunContinuationsAsynchronously);
try
{
Action handler = null;
handler = () =>
{
tcs.TrySetResult(true);
OnSendFromAudioStreamComplete -= handler;
};
OnSendFromAudioStreamComplete += handler;
InitialiseSendAudioFromStreamTimer(audioStream, streamSampleRate);
_streamSourceTimer.Change(_audioSamplePeriodMilliseconds, _audioSamplePeriodMilliseconds);
}
catch (Exception e)
{
Console.WriteLine(e.Message);
}
return tcs.Task;
}
}
else
{
return Task.CompletedTask;
}
}
Hello Master,
I need help to understand why I have this exception, it ends my application, what I see is that it generates when the audio is sent!
Error 9/13/2021 12:49:12 PM Application Error 1000 (100)
Application: UserAgentServer.exe CoreCLR Version: 4.700.20.20201 .NET Core Version: 3.1.4 Description: The process was terminated due to an unhandled exception. Exception Info: System.ObjectDisposedException: Cannot access a closed file. at System.IO.FileStream.get_Position() at SIPSorcery.Media.AudioExtrasSource.GetPcmSampleFromReader(BinaryReader binaryReader, AudioSamplingRatesEnum inputSampleRate, Int32& samplesRead) at SIPSorcery.Media.AudioExtrasSource.SendStreamSample(Object state) at System.Threading.ExecutionContext.RunFromThreadPoolDispatchLoop(Thread threadPoolThread, ExecutionContext executionContext, ContextCallback callback, Object state) --- End of stack trace from previous location where exception was thrown --- at System.Threading.ExecutionContext.RunFromThreadPoolDispatchLoop(Thread threadPoolThread, ExecutionContext executionContext, ContextCallback callback, Object state) at System.Threading.TimerQueueTimer.CallCallback(Boolean isThreadPool) at System.Threading.TimerQueueTimer.Fire(Boolean isThreadPool) at System.Threading.TimerQueueTimer.System.Threading.IThreadPoolWorkItem.Execute() at System.Threading.ThreadPoolWorkQueue.Dispatch() at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback()