FT2000RC Remote Audio is a Voice over IP (VoIP) solution primarily designed for two-way amateur radio remote control over the Internet. It uses streaming-audio technology in a client–server architecture. It features high radio sound quality, low latency, low bandwidth and very low CPU usage. FT2000RC Remote Audio uses the TCP protocol for an easy installation behind routers and firewalls. More than one user can connect to one server, the radio audio is broadcasted to all connected users. If you want to learn more about the concept and the difficulties read this article.
Once you have set up Remote Audio Server, the Connection to the Remote Audio Server and the Remote Audio Client settings, you are ready to go. If you have set your instance of FT2000RC as client, you will notice audio controls in the status bar of the application.
|Connect button||You can connect to the FT2000RC remote server by clicking on this button and you can disconnect also.|
|Connection status indicator||The status indicator can have three states or colors. Maroon indicates you are not connected. Lime indicates you are connected. Orange indicates a connection problem, e.g. the connection was rejected or the server is not reachable. If you hold the mouse over the indicator a hint shows you the description of the state.|
|Avg. latency||This value shows displays the average round trip latency of your internet connection to your server. Read more about that in the chapter Remote Connection Monitor.|
|Connect audio button||You can connect the remote audio server by clicking on this button and you can disconnect also. The server is the same of course. However, the audio is transferred over a separate TCP connection and port. You can configure your software so that you can connect the audio at the same time when you click on the Connect button. See Remote Audio Client settings.|
|Mic mute||It doesn't make any sense to send your own audio to the server when you are just listening to the receiver or work CW. Therefore your voice is only transmitted in voice modes (for example SSB) and the Microphone button is only enabled if you operate on MOX or VOX control. Check the MOX and or VOX checkboxes accordingly in the Remote Audio Client settings.|
|Receiver A button
Receiver B button
|If you receive dual (both FT-2000 or FT DX 5000 receivers on) set your Codec to stereo. You will be able to listen on both receivers. You can swap receiver A and receiver B audio by clicking on the A and B buttons.|
|Audio volume||The volume slider controls the audio volume on your local computer.|
|Audio transfer rate||This value shows the current average received audio transfer rate in kilobyte per second. The rate depends on the Codec settings (sample rate, bits per second and number of channels).|
Latency, glitches or interruptions in the audio transfer
To understand the difficulties associated with VoIP over Internet, one must first understand latency and how it works. Latency is the time delay between two ends of a VoIP line. It can be measured either one-way or round trip. A one-way latency of 120 milliseconds is acceptable for radio remote communication. A round-trip latency of over 300 milliseconds is considered poor. Many factors and components add latency to the audio path. Some can be reduced or optimised, others not. Surprisingly, most of the latency can be generated on your own computer and not on the Internet connection itself. One of the difficulties are buffers.
If the buffers are too small and the data runs out before Windows can get back to top them up (playback) or empty them (recording) you'll get a gap in the audio stream that sounds like a click or pop in the waveform and is often referred to as a 'glitch'. If the buffers are far too small, these glitches occur more often, firstly giving rise to occasional crackles and eventually to almost continuous interruptions that sound like distortion as the audio starts to break up regularly.
FT2000RC Remote Audio uses very small sized buffers for audio transfer to achieve minimum latency. You might experience glitches or interruptions in the audio transfer sometimes. This can have several reasons. One can be a busy Internet line in the evenings when everybody is at home and on the Internet. One suggestion is to switch to a Codec setting with a lower sample rate (8 kHz) and a bit rate of 8 bps.
The next advice is to open Windows Task Manager and to watch the CPU usage.
With several other applications open (e.g. Internet browser) and FT2000RC connected to the server the CPU usage should be 4% (sample above for a XP computer with 4GB RAM). When you open the audio connection to the server the CPU usage climbs to plus minus 9%. If you experience audio transfer interruptions watch the CPU usage again. If the CPU usage is 100% or almost 100% close other applications. Internet browsers for example can sometimes comsume many resources. If the interruptions are not gone re-boot the computer of the remote station.
Most of the latency on the VoIP path from your radio to your ears is however generated on your own computer! One major source of latency is the soundcard. The audio stream arrives in many small chunks of data. The soundcard needs time to process the data and to convert it from digital to analogue. If the audio stream packets arrive in a higher frequency than the soundcard can process them, the audio stream is buffered and the sound output delayed. This can lead to delays of 500ms or even more than one second and that is unacceptable for remote radio audio.
The soundcard buffer overrun (see Settings > Remote Audio - Client Codecs) can be limited and the latency reduced. This value can always be changed at runtime.
Another option to tune and optimize is the buffer length of the audio server as explained in the Server Audio Codecs.
RemAud - Remote Audio Client and Server
The FT2000RC remote audio is also available as a separate stand-alone client-server solution. Please check df3cb.com/remaud/.
The FT2000RC audio client can connect to a RemAud server.