SIPSorcery: Manually connect RTPSession to Audio Source

program.cs

using System;
using System.Threading;
using System.Threading.Tasks;
using SIPSorcery.Media;
using SIPSorcery.Net;
using SIPSorcery.SIP;
using SIPSorcery.SIP.App;
using SIPSorceryMedia.Abstractions.V1;

namespace CustomAudioSession
{
    class Program
    {
        static async Task Main()
        {
            CancellationTokenSource exitCts = new CancellationTokenSource();

            var sipTransport = new SIPTransport();
            var userAgent = new SIPUserAgent(sipTransport, null);
            userAgent.ClientCallFailed += (uac, error, sipResponse) => Console.WriteLine($"Call failed {error}.");
            userAgent.OnCallHungup += (dialog) => exitCts.Cancel();

            RTPSession rtpSession = new RTPSession(false, false, false);
            var audioTrack = new MediaStreamTrack(new AudioFormat(SDPWellKnownMediaFormatsEnum.PCMA));
            rtpSession.addTrack(audioTrack);

            var callResult = await userAgent.Call("[email protected]:6060", null, null, rtpSession);

            if (callResult)
            {
                Console.WriteLine("Call Answered.");

                AudioExtrasSource dummySource = new AudioExtrasSource();
                dummySource.SetAudioSourceFormat(rtpSession.GetSendingFormat(SDPMediaTypesEnum.audio).ToAudioFormat());
                dummySource.OnAudioSourceEncodedSample += rtpSession.SendAudio;
                dummySource.SetSource(AudioSourcesEnum.Music);

                Console.WriteLine("press ctrl-c to exit...");
                exitCts.Token.WaitHandle.WaitOne();
            }
            else
            {
                Console.WriteLine("Call Failed.");
            }
        }
    }
}