The video sampling from OnVideoSourceEncodedSample is jerky. as FrameArrivedHandler is not called at fixed intervals(the event is not raised for a long time. and all of a sudden, it is called a few times). This is a silly question that probably is non-relevant to SIPSorceryMedia.Windows project. I am new to video programming, is it caused by intensive cpu usage?

Any way to run it with Win7/Win8/Win8.1？
Other packages can use witth netstandard but this can not.

<TargetFramework>net6.0-windows10.0.22000</TargetFramework>

Shall we use NAudio for audio? or MediaFoundatiion for both video/video?

PauseAudio() and ResumeAudio() are only pausing _waveInEvent in WindowsAudioEndPoint.

        public Task PauseAudio()
        {
            _isPaused = true;
            _waveInEvent?.StopRecording();
            return Task.CompletedTask;
        }

        public Task ResumeAudio()
        {
            _isPaused = false;
            _waveInEvent?.StartRecording();
            return Task.CompletedTask;
        }

What if I want to pause _waveOutEvent as well?

does it make sense to do a PR, adding an option for that (that defaults to the old behaviour)? like this:

        public Task PauseAudio(bool affectWaveOut = false)
        {
            _isPaused = true;
            _waveInEvent?.StopRecording();
		if(affectWaveOut)
		{
			_waveOutEvent?.Pause();
		}

            return Task.CompletedTask;
        }

        public Task ResumeAudio(bool affectWaveOut = false)
        {
            _isPaused = false;
            _waveInEvent?.StartRecording();
		if(affectWaveOut)
		{
			_waveOutEvent?.Play();
		}
            return Task.CompletedTask;
        }

Or am I missing something?

During the process of sending the video, decoding failed and the program crashed. The following is the error stack information:

System.ApplicationException: VP8 encode attempt failed, Unspecified internal error.
at SIPSorceryMedia.Encoders.Codecs.Vp8Codec.Encode(Byte[] frame, VpxImgFmt inputPixelFormat, Boolean forceKeyFrame)
at SIPSorceryMedia.Encoders.VpxVideoEncoder.EncodeVideo(Int32 width, Int32 height, Byte[] sample, VideoPixelFormatsEnum pixelFormat, VideoCodecsEnum codec)
at SIPSorceryMedia.Windows.WindowsVideoEndPoint.FrameArrivedHandler(MediaFrameReader sender, MediaFrameArrivedEventArgs e)
at System.Threading.Tasks.Task.<>c.b__128_1(Object state)
at System.Threading.ThreadPoolWorkQueue.Dispatch()
at System.Threading.PortableThreadPool.WorkerThread.WorkerThreadStart()
at System.Threading.Thread.StartCallback()

Hi there
How can I use lib with net 7.0?

When I use WebRTC for audio and video communication, the audiodg.exe program takes up too much CPU, causing the program to crash. What could be the reason? The picture above shows my system version, CPU, memory and other information. Looking forward to hearing from you!

Part of the code is as follows:
WindowsAudioEndPoint _winAudioEP = new WindowsAudioEndPoint(new AudioEncoder());
RTCConfiguration config = new() { iceServers = _iceServers, X_UseRtpFeedbackProfile = true };
_pc = new RTCPeerConnection(config);
MediaStreamTrack audioTrack = new(_winAudioEP.GetAudioSinkFormats(), MediaStreamStatusEnum.SendRecv);
_pc.addTrack(audioTrack);
_pc.OnAudioFormatsNegotiated += (audioFormats) =>
{
AudioVideoSendService.AudioFormat = audioFormats.First();
_winAudioEP.SetAudioSinkFormat(audioFormats.First());
};
_pc.OnRtpPacketReceived += (IPEndPoint rep, SDPMediaTypesEnum media, RTPPacket rtpPkt) =>
{
if (media == SDPMediaTypesEnum.audio)
{
_winAudioEP.GotAudioRtp(rep, rtpPkt.Header.SyncSource, rtpPkt.Header.SequenceNumber, rtpPkt.Header.Timestamp, rtpPkt.Header.PayloadType, rtpPkt.Header.MarkerBit == 1, rtpPkt.Payload);
}
};

System.Runtime.InteropServices.COMException (0xC00D36B2)
at WinRT.ExceptionHelpers.g__Throw|39_0(Int32 hr)
at ABI.Windows.Media.Capture.Frames.IMediaFrameReaderMethods.StopAsync(IObjectReference _obj)
at SIPSorceryMedia.Windows.WindowsVideoEndPoint.CloseVideoCaptureDevice()
at SIPSorceryMedia.Windows.WindowsVideoEndPoint.VideoCaptureDevice_Failed(MediaCapture sender, MediaCaptureFailedEventArgs errorEventArgs)
at System.Threading.Tasks.Task.<>c.b__128_1(Object state)
at System.Threading.ThreadPoolWorkQueue.Dispatch()
at System.Threading.PortableThreadPool.WorkerThread.WorkerThreadStart()

target framework net472

Depending on the device I am trying to call, I can either only hear what the person on the other side is saying, or only they can hear what I am saying. I am trying to call "regular" telephone numbers (via my AVM Fritz Box as a SIP server). My wife on her Iphone and my colleage on the landline at work can talk to me but not hear me, my brother on his android phone can hear me (sort of) but not talk to me.

Hello all, I'm fairly new to the world of SIP and I'm running into a bit of a bind with this library. My goal is fairly straightforward, I want to create a little .NET service that sends audio to SIP endpoints. I'm trying to use a WindowsAudioEndpoint declared like it is in the SIPExamples project Play Sounds.

var windowsAudio = new WindowsAudioEndPoint(new AudioEncoder());
windowsAudio.RestrictFormats(format => format.Codec == AudioCodecsEnum.G722);
var voipMediaSession = new VoIPMediaSession(windowsAudio.ToMediaEndPoints());

This is a result of trying to improve the audio quality produced by a VoIPMediaSession declared like this:

VoIPMediaSession voipMediaSession = new VoIPMediaSession(null, formats => formats.Codec == AudioCodecsEnum.G722);

The audio coming out of my softphone and speakers is much more consistent and clear with the former declaration rather than the latter. The latter is basically unlistenable for some reason?

Now the problem is, and this is a problem even running the stock PlaySounds example against my endpoints, that an exception is thrown while the call task is awaited.
The debugger takes me to line 141 of WindowsAudioEndPoint.cs:
if (!_disableSource && _waveSourceFormat.SampleRate != _audioFormatManager.SelectedFormat.ClockRate)
where it seems like _waveSourceFormat is null.

The naïve fix I'm using is to add a null conditional operator to said _waveSourceFormat on line 141 and on line 143.

Any help or advice would be much appreciated, thanks!

I get this error message when trying to add the SIPSorceryMedia.Windows nuget package to my .net6 project, latest windows 11 SDK installed.

"NU1202 Package SIPSorceryMedia.Windows 6.0.1-pre is not compatible with net6.0 (.NETCoreApp,Version=v6.0). Package SIPSorceryMedia.Windows 6.0.1-pre supports: net6.0-windows10.0.22000 (.NETCoreApp,Version=v6.0)"

Any Ideas?

Currently WindowsAudioEndPoint class uses the hardcoded values of the channels number and the sample rate for source and sink devices:

private const int DEVICE_CHANNELS = 1;
...
public readonly static AudioSamplingRatesEnum DefaultAudioSourceSamplingRate = AudioSamplingRatesEnum.Rate8KHz;
public readonly static AudioSamplingRatesEnum DefaultAudioPlaybackRate = AudioSamplingRatesEnum.Rate8KHz;

Is it done on purpose? Why do we stick to minimal values if we can determine the devices capabilities?

For example, on my PC I have captureChannels=1, captureSampleRate=48000 and renderChannels=2, renderSampleRate=48000 (see code below).

NAudio.CoreAudioApi.MMDeviceEnumerator enumerator = new NAudio.CoreAudioApi.MMDeviceEnumerator();

NAudio.CoreAudioApi.MMDevice captureDevice = enumerator.GetDefaultAudioEndpoint(
    NAudio.CoreAudioApi.DataFlow.Capture, NAudio.CoreAudioApi.Role.Communications);
var captureChannels = captureDevice.AudioClient.MixFormat.Channels;
var captureSampleRate = captureDevice.AudioClient.MixFormat.SampleRate;

NAudio.CoreAudioApi.MMDevice renderDevice = enumerator.GetDefaultAudioEndpoint(
    NAudio.CoreAudioApi.DataFlow.Render, NAudio.CoreAudioApi.Role.Communications);
var renderChannels = renderDevice.AudioClient.MixFormat.Channels;
var renderSampleRate = renderDevice.AudioClient.MixFormat.SampleRate;