public void Stop()
{
// Stop the keepalive timer
if (keepalive_timer != null)
{
keepalive_timer.Stop();
keepalive_timer = null;
}
if (rtsp_client == null)
{
return;
}
// Send TEARDOWN
var teardown_message = new RtspRequestTeardown {
RtspUri = new Uri(Url),
Session = RtspSession
};
if (auth_type != null)
{
AddAuthorization(teardown_message, Username, Password, auth_type, realm, nonce, Url);
}
rtsp_client.SendMessage(teardown_message);
// Drop the RTSP session
rtsp_client.Stop();
}
private void RTSP_ProcessTeardownRequest(RtspRequestTeardown message, RtspListener listener)
{
if (message.Session == _videoSessionId) // SHOULD HAVE AN AUDIO TEARDOWN AS WELL
{
// If this is UDP, close the transport
// For TCP there is no transport to close (as RTP packets were interleaved into the RTSP connection)
Rtsp.Messages.RtspResponse teardown_response = message.CreateResponse(_logger);
listener.SendMessage(teardown_response);
CloseConnection("teardown");
}
}
internal RtspResponse HandleTeardown(RtspRequestTeardown request)
{
Contract.Requires(request != null);
Contract.Ensures(Contract.Result <RtspResponse>() != null);
var response = request.CreateResponse();
RtspPushDescription description;
if (!PushDescriptions.TryGetValue(request.RtspUri.AbsolutePath, out description))
{
response.ReturnCode = 404;
return(response);
}
description.Stop(request.Session);
return(response);
}
public void Stop()
{
if (rtspListener != null)
{
RtspRequest teardownMessage = new RtspRequestTeardown
{
RtspUri = new Uri(url),
Session = session
};
rtspListener.SendMessage(teardownMessage);
}
// clear up any UDP sockets
udpPair?.Stop();
// Stop the keepalive timer
timer?.Dispose();
// Drop the RTSP session
rtspListener?.Stop();
}
