Trouble with creating an audio endpoint

[tilkinsc]

Hi, the documentation for this library seems to be leagues out of date so I was going with examples. This library is also not published on nuget with sufficient sources?, as it lacks source information so my LSP can't really walk through more than decompiling whatever file you are on so jumping around is a nightmare/impossible.

I am using voip.ms as my SIP people and I was trying to dial their *225 number to say my account balance. I was able to confirm everything is working up to the receiving call portion which doesn't 403. However, I don't see any events firing off in my audio events. My audio driver works with the iptel music example. Voip.ms supports G711 and G722 fully, which I have enabled and I have tested that I can make a call with a pre-existing softphone software. I have encryption disabled. I expect that my audio output event from NAudio to receive data, but it doesn't fire off at all.

Can anyone spot what I am doing wrong?

* Note: my username/password/server is correct

// NAudioPCMEndpoint.cs
namespace Sippy.Audio;

class NAudioPCMEndpoint : IAudioSource, IAudioSink
{
	public event EncodedSampleDelegate? OnAudioSourceEncodedSample;
	public event RawAudioSampleDelegate? OnAudioSourceRawSample;
	public event SourceErrorDelegate? OnAudioSourceError;
	public event SourceErrorDelegate? OnAudioSinkError;
	
	public bool HasEncodedAudioSubscribers() => OnAudioSourceEncodedSample != null;
	
	private MediaFormatManager<AudioFormat> _mediaFormatManager;
	public List<AudioFormat> GetAudioSinkFormats() => _mediaFormatManager.GetSourceFormats();
	public List<AudioFormat> GetAudioSourceFormats() => _mediaFormatManager.GetSourceFormats();
	public void RestrictFormats(Func<AudioFormat, bool> filter) => _mediaFormatManager.RestrictFormats(filter);
	public void SetAudioSinkFormat(AudioFormat audioFormat) => _mediaFormatManager.SetSelectedFormat(audioFormat);
	public void SetAudioSourceFormat(AudioFormat audioFormat) => _mediaFormatManager.SetSelectedFormat(audioFormat);
	
	private IAudioEncoder _audioEncoder;
	
	private readonly WaveInEvent? _waveIn;
	private readonly WaveOutEvent? _waveOut;
	private readonly BufferedWaveProvider _waveProvider;
	
	private bool _isStarted = false;
	private bool _isClosed = false;
	private bool _isAudioSourcePaused = false;
	public bool IsAudioSourcePaused() => _isAudioSourcePaused;
	
	public NAudioPCMEndpoint(IAudioEncoder audioEncoder)
	{
		_audioEncoder = audioEncoder;
		_mediaFormatManager = new(audioEncoder.SupportedFormats);
		AudioCodecsEnum[] codecs = [ AudioCodecsEnum.G722 ];
		_mediaFormatManager.RestrictFormats(format => codecs.Contains(format.Codec));
		
		WaveFormat inputWaveFormat = new(8000, 16, 1);
		_waveIn = new()
		{
			WaveFormat = inputWaveFormat,
			NumberOfBuffers = 2
		};
		_waveIn.DataAvailable += InputDataAvailable;
		
		WaveFormat outputWaveFormat = new(16000, 16, 1);
		_waveProvider = new(outputWaveFormat);
		_waveOut = new();
		_waveOut.Init(_waveProvider);
	}
	
	private void InputDataAvailable(object? sender, WaveInEventArgs e)
	{
		short[] samples = new short[e.BytesRecorded / 2];
		for (int index = 0; index < samples.Length; index++)
		{
			samples[index] = BitConverter.ToInt16(e.Buffer, index * 2);
		}

		byte[] encodedSample = _audioEncoder.EncodeAudio(samples, _mediaFormatManager.SelectedFormat);
		OnAudioSourceEncodedSample?.Invoke((uint)encodedSample.Length, encodedSample);
	}
	
	public void ExternalAudioSourceRawSample(AudioSamplingRatesEnum samplingRate, uint durationMilliseconds, short[] sample)
	{
		throw new NotImplementedException();
	}

	public void GotAudioRtp(IPEndPoint remoteEndPoint, uint ssrc, uint seqnum, uint timestamp, int payloadID, bool marker, byte[] payload)
	{
		// does not fire
		Console.WriteLine("Audio RTP");
		short[] pcmSample = _audioEncoder.DecodeAudio(payload, _mediaFormatManager.SelectedFormat);
		
		var sampleBytes = pcmSample.SelectMany(x => new byte[] { (byte) (x >> 8), (byte) x }).ToArray();
		
		_waveProvider.AddSamples(sampleBytes, 0, sampleBytes.Length);
	}

	public Task StartAudio()
	{
		if (!_isStarted)
		{
			_isStarted = true;
			_waveIn?.StartRecording();
			_waveOut?.Play();
		}
		
		return Task.CompletedTask;
	}
	
	public Task StartAudioSink() => StartAudio();

	public Task CloseAudio()
	{
		if (!_isClosed)
		{
			_isClosed = true;
			_waveIn?.StopRecording();
			_waveIn?.Dispose();
			_waveOut?.Stop();
			_waveOut?.Dispose();
		}
		
		return Task.CompletedTask;
	}

	public Task CloseAudioSink() => CloseAudio();
	
	public Task PauseAudio()
	{
		_isAudioSourcePaused = true;
		_waveIn?.StopRecording();
		return Task.CompletedTask;
	}

	public Task PauseAudioSink() => PauseAudio();

	public Task ResumeAudio()
	{
		_isAudioSourcePaused = false;
		_waveIn?.StartRecording();
		return Task.CompletedTask;
	}

	public Task ResumeAudioSink() => ResumeAudio();

	public MediaEndPoints ToMediaEndPoints()
	{
		return new MediaEndPoints
		{
			AudioSource = this,
			AudioSink = this
		};
	}
}

namespace Sippy.Call;

class SIPManager
{
	
	private const int SIP_LISTEN_PORT = 5060;
	private const int SIP_WEBSOCKET_LISTEN_PORT = 80;
	
	private SIPTransport? _sipTransport;
	private SIPRegistrationUserAgent? _registrationUserAgent;
	
	public bool IsLoggedIn { get; private set; }
	
	public void OpenTransport()
	{
		if (_sipTransport is null)
		{
			_sipTransport = new();
			_sipTransport.EnableTraceLogs();
			_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
			_sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.Any, SIP_LISTEN_PORT)));
			_sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.Any, SIP_WEBSOCKET_LISTEN_PORT));
			_sipTransport.AddSIPChannel(new SIPUDPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
			_sipTransport.AddSIPChannel(new SIPTCPChannel(new IPEndPoint(IPAddress.IPv6Any, SIP_LISTEN_PORT)));
			_sipTransport.AddSIPChannel(new SIPWebSocketChannel(IPAddress.IPv6Any, SIP_WEBSOCKET_LISTEN_PORT));
		}
	}
	
	public void CloseTransport()
	{
		_sipTransport?.Shutdown();
		_sipTransport?.Dispose();
		_sipTransport = null;
	}
	
	public async Task LoginAsync()
	{
		if (_sipTransport is not null)
		{
			TaskCompletionSource tcs = new();
			
			SIPRegistrationUserAgent registrationUserAgent = new(
				_sipTransport,
				"sipusername",
				"sippassword",
				"server",
				30
			);
			registrationUserAgent.RegistrationFailed += (uri, resp, err) => {
				IsLoggedIn = false;
				tcs.TrySetResult();
			};
			registrationUserAgent.RegistrationTemporaryFailure += (uri, resp, msg) => {
				IsLoggedIn = false;
				tcs.TrySetResult();
			};
			registrationUserAgent.RegistrationRemoved += (uri, resp) => {
				IsLoggedIn = false;
				tcs.TrySetResult();
			};
			registrationUserAgent.RegistrationSuccessful += (uri, resp) => {
				IsLoggedIn = true;
				tcs.TrySetResult();
				Console.WriteLine("Logged in");
				Console.WriteLine(registrationUserAgent.UserDisplayName);
			};
			
			registrationUserAgent.Start();
			
			// force login sync in case I want to test anything that requires it
			await tcs.Task;
			_registrationUserAgent = registrationUserAgent;
		}
	}
	
	public void Logout()
	{
		_registrationUserAgent?.Stop(true);
		IsLoggedIn = false;
	}
	
	public async Task MakeOutboundCallTo(string sipUri)
	{
		if (_registrationUserAgent is not null)
		{
			// Only report supporting G722
			List<AudioCodecsEnum> codecs = [AudioCodecsEnum.G722];
			
			var userAgent = new SIPUserAgent(_sipTransport, null, true);
			var audioEncoder = new AudioEncoder();
			var naudioEndpoint = new NAudioPCMEndpoint(audioEncoder);
			var voipMediaSession = new VoIPMediaSession(naudioEndpoint.ToMediaEndPoints())
			{
				AcceptRtpFromAny = true
			};
			voipMediaSession.OnRtpPacketReceived += (ep, type, rtp) => {
				Console.WriteLine($"{ep} {type} {rtp}");
			};
			voipMediaSession.OnAudioFormatsNegotiated += (formats) => {
				formats.ForEach(format => Console.WriteLine(format.Codec));
			};
			voipMediaSession.OnReceiveReport += (mediaType, packet, report) => {
				Console.WriteLine($"{mediaType} {packet} {report}");
			};
			userAgent.OnCallHungup += (dialogue) => {
				Console.WriteLine("Disconnected");
			};
			userAgent.OnRtpEvent += (rtpevent, rtpheader) => {
				Console.WriteLine($"{rtpevent} : {rtpheader}");
			};
			userAgent.RemotePutOnHold += () => {
				Console.WriteLine("We are on hold");
			};
			userAgent.ClientCallAnswered += (ISIPClientUserAgent uac, SIPResponse sipResponse) => {
				Console.WriteLine("Client call answered");
			};
			userAgent.OnDtmfTone += (byte key, int duration) => {
				string callId = userAgent.Dialogue.CallId;
				Console.WriteLine($"Call {callId} received DTMF tone {key}, duration {duration}ms.");
			};
			
			bool callResult = await userAgent.Call(sipUri, "sipusername", "sippassword", voipMediaSession);
			Console.WriteLine($"Call result {(callResult ? "success" : "failure")}.");
			
			Console.WriteLine("press any key to exit...");
			Console.Read();
			
			if (userAgent.IsCallActive)
			{
				Console.WriteLine("Hanging up.");
				userAgent.Hangup();
				
				await Task.Delay(1000);
			}
		}
	}
	
}

// Program.cs
class Program
{
	
	private static string DESTINATION = "sip:*225@server"; // has correct server address
	
	public static async Task Main(string[] args)
	{
		SIPManager manager = new();
		manager.OpenTransport();
		await manager.LoginAsync();
		await manager.MakeOutboundCallTo(DESTINATION);
		manager.Logout();
		manager.CloseTransport();
	}
	
}

[tilkinsc]

I think the Softphone example has what I need I'm gonna try to program around that one. Perhaps I misunderstood the roles of the classes.