[WRN] SendRtpRaw was called for a "audio" packet on a closed RTP session.

[calebtt]

Something is persisting in the background after a call is answered, hung up, and then a new call is answered. It generates a lot of warning logs like these. The audio end point doesn't really have a way to discern if the RTP session it's trying to forward to is "closed" and as far as my code goes I don't think there should be anything left around after the first call ends.

It's probably some part of the Sip Sorcery stack I am using wrong.

So my question is, does anything in my SipClient stand out as obviously wrong for idling to handle incoming calls that would cause the SendRtpRaw warning?

2025-05-22 12:37:28.426 -05:00 [WRN] SendRtpRaw was called for a "audio" packet on a closed RTP session.
2025-05-22 12:37:28.427 -05:00 [DBG] [LocalVoiceAudioEndPoint] Forwarded encoded sample: 160 bytes, 160 RTP units

Here are the relevant parts of my sip client code,

using Serilog;
using SIPSorcery.Media;
using SIPSorcery.Net;
using SIPSorcery.SIP;
using SIPSorcery.SIP.App;
using SIPSorceryMedia.Abstractions;

public class SipClient
{
    private const int RegistrationExpirySeconds = 60;

    private readonly string _sipUsername;
    private readonly string _sipPassword;
    private readonly string _sipServer;
    private readonly string _sipFromName;

    private readonly MediaEndPoints _mediaEndPoints;

    private SIPTransport _sipTransport;
    private SIPUserAgent _userAgent;
    private SIPServerUserAgent? _pendingIncomingCall;
    private CancellationTokenSource _cts = new();

    public event Action<SipClient>? CallAnswer;
    public event Action<SipClient>? CallEnded;
    public event Action<SipClient, string>? StatusMessage;
    public event Action<SipClient>? RemotePutOnHold;
    public event Action<SipClient>? RemoteTookOffHold;

    private VoIPMediaSession? _mediaSession = null;
    private SIPRegistrationUserAgent _registrationAgent;

    // CTOR
    public SipClient(
        SIPTransport sipTransport,
        MediaEndPoints mediaEndPoints,
        SipConfig sipSettings)
    {
        _sipTransport = sipTransport;
        _sipUsername = sipSettings.Username;
        _sipPassword = sipSettings.Password;
        _sipServer = sipSettings.Server;
        _sipFromName = sipSettings.FromName;

        //m_audioOutDeviceIndex = audioOutDeviceIndex;
        _mediaEndPoints = mediaEndPoints ?? throw new ArgumentNullException(nameof(mediaEndPoints));
        if (mediaEndPoints.AudioSink == null || mediaEndPoints.AudioSource == null)
        {
            throw new ArgumentException("AudioSink and AudioSource must be provided in MediaEndPoints.", nameof(mediaEndPoints));
        }

        _sipTransport.SIPRequestInTraceEvent += (localSIPEndPoint, endPoint, request) =>
            Log.Debug($"SIP Request Received: {request.Method} from {endPoint}");

        _sipTransport.SIPResponseOutTraceEvent += (localSIPEndPoint, endPoint, response) =>
            Log.Debug($"SIP Response Sent: {response.Status} to {endPoint}");

        StunHelper.SetupStun();

        _userAgent = new SIPUserAgent(_sipTransport, null);
        _userAgent.ClientCallTrying += CallTrying;
        _userAgent.ClientCallRinging += CallRinging;
        _userAgent.ClientCallAnswered += CallAnswered;
        _userAgent.ClientCallFailed += CallFailed;
        _userAgent.OnCallHungup += CallFinished;
        _userAgent.ServerCallCancelled += IncomingCallCancelled;

        // Initialize registration agent
        _registrationAgent = new SIPRegistrationUserAgent(
            sipTransport,
            _sipUsername,
            _sipPassword,
            _sipServer,
            RegistrationExpirySeconds
        );

        _registrationAgent.RegistrationSuccessful += (uri, resp) =>
StatusMessage?.Invoke(this, $"Registration successful for {uri}. Expires: {resp.Header.Expires}");
        _registrationAgent.RegistrationFailed += (uri, resp, error) =>
            StatusMessage?.Invoke(this, $"Registration failed for {uri}: {error}");
        _registrationAgent.RegistrationTemporaryFailure += (uri, resp, error) =>
            StatusMessage?.Invoke(this, $"Registration temporary failure for {uri}: {error}");

    }

    public void StartRegistration()
    {
        _registrationAgent.Start();
        StatusMessage?.Invoke(this, $"Registration attempt for {_sipUsername}@{_sipServer} started.");
    }

    public void Shutdown()
    {
        Hangup();
        _mediaSession?.Dispose();
        _mediaSession = null;
        _registrationAgent?.Stop();
        _sipTransport.Shutdown();
    }

    public void Accept(SIPRequest sipRequest)
    {
        _pendingIncomingCall = _userAgent.AcceptCall(sipRequest);
    }

    public async Task<bool> Answer()
    {
        if (_pendingIncomingCall == null)
        {
            StatusMessage?.Invoke(this, $"There was no pending call available to answer.");
            return false;
        }
        else
        {
            var sipRequest = _pendingIncomingCall.ClientTransaction.TransactionRequest;

            bool hasAudio = true;
            bool hasVideo = false;

            if (sipRequest.Body != null)
            {
                SDP offerSDP = SDP.ParseSDPDescription(sipRequest.Body);
                //offerSDP.AddressOrHost = StunHelper.PublicIPAddress;
                hasAudio = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.audio && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
                hasVideo = offerSDP.Media.Any(x => x.Media == SDPMediaTypesEnum.video && x.MediaStreamStatus != MediaStreamStatusEnum.Inactive);
            }

            _mediaSession?.Dispose();
            _mediaSession = CreateMediaSession();

            bool result = await _userAgent.Answer(_pendingIncomingCall, _mediaSession);
            _pendingIncomingCall = null;

            if (result)
            {
                CallAnswer?.Invoke(this);
            }

            return result;
        }
    }

    public void Hangup()
    {
        if (_userAgent.IsCallActive)
        {
            _userAgent.Hangup();
            CallFinished(null);
            CallEnded?.Invoke(this);
        }
    }

    private VoIPMediaSession CreateMediaSession()
    {
        var voipMediaSession = new VoIPMediaSession(_mediaEndPoints);
        //voipMediaSession.AcceptRtpFromAny = false;
        Log.Information($"[{GetType().Name}] Created with AudioSink={_mediaEndPoints.AudioSink.GetType().Name}, AcceptRtpFromAny={voipMediaSession.AcceptRtpFromAny}");
        return voipMediaSession;
    }

    private void CallTrying(ISIPClientUserAgent uac, SIPResponse sipResponse)
    {
        StatusMessage?.Invoke(this, "Call trying: " + sipResponse.StatusCode + " " + sipResponse.ReasonPhrase + ".");
    }

    private void CallRinging(ISIPClientUserAgent uac, SIPResponse sipResponse)
    {
        StatusMessage?.Invoke(this, "Call ringing: " + sipResponse.StatusCode + " " + sipResponse.ReasonPhrase + ".");
    }

    private void CallFailed(ISIPClientUserAgent uac, string errorMessage, SIPResponse? failureResponse)
    {
        StatusMessage?.Invoke(this, "Call failed: " + errorMessage + ".");
        CallFinished(null);
    }

    private void CallAnswered(ISIPClientUserAgent uac, SIPResponse sipResponse)
    {
        StatusMessage?.Invoke(this, "Call answered: " + sipResponse.StatusCode + " " + sipResponse.ReasonPhrase + ".");
        CallAnswer?.Invoke(this);
    }

    private void CallFinished(SIPDialogue? dialogue)
    {
        _mediaSession?.Dispose();
        _mediaSession = null;
        _pendingIncomingCall = null;
        CallEnded?.Invoke(this);
    }

    private void IncomingCallCancelled(ISIPServerUserAgent uas, SIPRequest cancelRequest)
    {
        CallFinished(null);
    }
}

[sipsorcery]

It'd be worth trying _mediaSession.Close in your CallFinished method.

[calebtt]

I looked into that, the dispose() method calls that with it's own reason string.