I am new in the .Net and, especially, in the SIP space.

I am using the SIPSorcery nuget package to create a SIP softphone of sorts. I want to receive a call and record it. This is my code:

static async Task Main()
{
SIPTransport _sipTransport = new SIPTransport();

SIPRegistrationUserAgent _userAgent = new(_sipTransport, SIP_USERNAME, SIP_PASSWORD, SIP_SERVER, DEFAULT_EXPIRY);

_userAgent.RegistrationSuccessful += (uri, resp) =>
{
    Console.WriteLine($"{uri} registration succeeded.");

    SIPUserAgent _ua = new SIPUserAgent(_sipTransport, null, true);

    _ua.OnIncomingCall += async (ua, req) =>
    {
        WindowsAudioEndPoint winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
        VoIPMediaSession voipMediaSession = new VoIPMediaSession(winAudioEP.ToMediaEndPoints());
        voipMediaSession.AcceptRtpFromAny = true;
        voipMediaSession.OnRtpPacketReceived += OnRtpPacketReceived;

        var uas = ua.AcceptCall(req);

        await ua.Answer(uas, voipMediaSession);
    };
};

_userAgent.Start();
}


private static void OnRtpPacketReceived(IPEndPoint remoteEndPoint, SDPMediaTypesEnum mediaType, RTPPacket rtpPacket)
{
    if (mediaType == SDPMediaTypesEnum.audio)
    {
        var sample = rtpPacket.Payload;

        for (int index = 0; index < sample.Length; index++)
        {
            if (rtpPacket.Header.PayloadType == (int)SDPWellKnownMediaFormatsEnum.PCMA)
            {
                short pcm = NAudio.Codecs.ALawDecoder.ALawToLinearSample(sample[index]);
                byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
                _waveFile.Write(pcmSample, 0, 2);
            }
            else
            {
                short pcm = NAudio.Codecs.MuLawDecoder.MuLawToLinearSample(sample[index]);
                byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };
                _waveFile.Write(pcmSample, 0, 2);
            }
        }
    }
}

The registration and the connection works. I do also get audio and save it in that _waveFile file.

However, the audio I get is noise. No real voice or anything like that. In the case of calling a sip address, I did not encounter such a problem. That is why I am confused. I would think I would have the problem when calling too if it was something that has to do with a Codec or something like that.