playing audio has a bad quality at client side

[JensNaue]

Hi all,
i need a program, that calls a group of phone clients (of a swyx-server) and plays an audio file to them, when connection has been established. For testing i used the "play audio" example. Everything works fine, but the audio sounds very fragmented.
My question is, if i use sipsorcery the right way, or should i try another approach? Thanks in advance for any suggestions.
Here's my code so far:
class Program
{
static async Task Main(string[] args)
{
var wavFile = @"C:\ProgramData....wav";
var dir = Path.GetDirectoryName(wavFile);
var name = Path.GetFileName(wavFile);
var newName = name.Replace(".wav", ".16kHz.wav");
var audioFile = Path.Combine(dir, newName);
if (File.Exists(audioFile) || ConvertWaveFile(wavFile, audioFile))
{
var sipTransport = new SIPTransport();

            var userAgent = new SIPUserAgent(sipTransport, null);
            var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());

            var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());
            voipMediaSession.AcceptRtpFromAny = true;
            voipMediaSession.OnAudioFormatsNegotiated += (formats) =>
                    voipMediaSession.AudioExtrasSource.SetAudioSourceFormat(formats.First());

            bool callResult = await userAgent.Call(
                "sip:<receiver-address>",
                "<sender name>",
                "<sender-pw>",
                voipMediaSession);

            if (callResult)
            {
                await windowsAudio.PauseAudio();
                try
                {
                    await voipMediaSession.AudioExtrasSource.StartAudio();

                    if (voipMediaSession.AudioRemoteTrack != null)
                    {
                        voipMediaSession.AudioRemoteTrack.MaximumBandwidth = 0;
                    }

                    await voipMediaSession.AudioExtrasSource.SendAudioFromStream(
                        new FileStream(audioFile, FileMode.Open, FileAccess.Read, FileShare.Read),
                        AudioSamplingRatesEnum.Rate16KHz);

                    voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.None);

                    await voipMediaSession.AudioExtrasSource.PauseAudio();

                    await Task.Delay(200);
                }
                catch (Exception ex)
                {
                    Console.WriteLine("error: " + ex.Message);
                }
                await windowsAudio.ResumeAudio();
                userAgent.Hangup();
            }
            ...
    static bool ConvertWaveFile(string source, string target)
    {
        if (!File.Exists(source))
            return false;
        try
        {
            using (var reader = new AudioFileReader(source))
            {
                var floatFormat = WaveFormat.CreateIeeeFloatWaveFormat(16000, 1);
                using (var resampler = new MediaFoundationResampler(reader, floatFormat))
                {
                    resampler.ResamplerQuality = 60;
                    var floatTo16 = new WaveFloatTo16Provider(resampler);

                    WaveFileWriter.CreateWaveFile(target, floatTo16);
                }
            }
            Console.WriteLine("Konvertierung erfolgreich abgeschlossen.");
        }
        catch (Exception ex)
        {
            Console.WriteLine($"Fehler bei der Konvertierung: {ex.Message}");
            return false;
        }
        return File.Exists(target);
    }

[sipsorcery]

What's the audio quality like with only a single call?

[JensNaue]

The test triggers only a single call yet. Aaron Clauson ***@***.***> schrieb am Fr. 7. März 2025 um 14:59:

…

What's the audio quality like with only a single call? — Reply to this email directly, view it on GitHub <#1352 (comment)>, or unsubscribe <https://github.com/notifications/unsubscribe-auth/BQHEQ5QBZVFSCRX2FG3RRX32TGQ4ZAVCNFSM6AAAAABYRI2LJOVHI2DSMVQWIX3LMV43URDJONRXK43TNFXW4Q3PNVWWK3TUHMYTENBSGY4TKNA> . You are receiving this because you authored the thread.Message ID: ***@***.*** com>

[sipsorcery]

In that case it's almost certainly a problem with audio file conversion or sampling rate. Try your example with a known good file.

[JensNaue]

Hi Aaron,
i tested it with all three files - at client side all of them sound choppy (just like an old robot ;) )
Of course i changed the second arg of the voipMediaSession.AudioExtrasSource.SendAudioFromStream function according to the sample rate of the file, that i used.
Server and client accept G722, G711 and G729.
What else can i try?
Thank you for further suggestions.
Jens

[sipsorcery]

I've just checked the PlaySounds and SIPCallServer (call 0@localhost to get the music file to play) demoes with Microsip and the audio is good for G711 and G722.

[calebtt]

I am having this same problem, how do I debug this?

[sipsorcery]

WireShark is the best way see this comment.

[calebtt]

Okay, I'll try to get it installed.

[calebtt]

My problem was an incorrect computation when sending an RTP frame.