Recording Multiple Inputs Over the 'Net?

TFGeditor asks: "Thanks to the advice of fellow readers from a previous Ask Slashdot, I now have a PC system optimally configured to produce professional on-air radio programs. Now I have a new problem: my radio co-host and I are in different cities located a few hundred miles apart. In order to give the show a real-time (i.e. 'live') sound, we need to somehow connect us so that we can produce a show complete with co-host banter, real-time interaction, and so on. I want it to sound as if we were both in the same studio. How can we do this? Will Skype or other VOIP applications do this without the result sounding 'tinny' (like a phone connection)? Are there other apps that will do a better job?"

Hi fidelity voiceIP

[TropicalCoder]

I developed an application that sends CD quality stereo audio over the internet in real time (one way connection). As input, it takes whatever audio is presented to the input of your sound card (which could be professional microphones, for example) and compresses it to 128 kb mp3 before sending via TCP or UDP packets. TCP requires at least 30% more bandwidth than UDP. For UDP, about 384 Kbits of bandwidth should do, while TCP may need up to 512 Kbits. In UDP mode, some UDP packets are returned to the sender to create a kind of handshaking to inform the sender that his packets are being received.

Audio is send four mp3 frames at a time, resulting in a latency of about one-tenth of a second for both send and receive. In UDP mode, there is the option of selecting some number of buffers so that the audio will be buffered to prevent drop out. Of course then lag will be multiplied by the number of buffers. On top of that you have the latency of your internet connection. Altogether, the lag could be quite acceptable if you have a good connection.

This application worked quite well in all my tests, but you could encounter issues with getting past a firewall or a DSL router/modem. Nothing in the software deals with these issues. I would be willing to "permanently lend" this application to you to experiment with, but you would need a certain level of technical knowledge to get past your router/modem/firewall. To use this application, you would mix the incoming signal from your partner with you own voice and music. Your partner of course will be monitoring the show. This software requires DirectX.

I developed this to teach myself about winsock. I don't know if there is any future in this software since it does not employ the RTP protocol for audio transmission and RTSP for audio signaling like a typical VOIP app, and it depends on mp3. However, it works very solidly and efficiently. I have thoroughly tested it both via the loopback on my computer, and over a computer network, with both TCP and UDP. I never managed to find someone capable of helping me test it over the internet. I would be happy to give a copy of this software to anyone wanting to experiment with it, and especially with anyone that has more than my minimal knowledge about resolving these issues like getting past the router/modem/firewall. If some other programmer thinks this may have a future and would like to colaborate with me on some project, that would be great. You will find my email address on my web site - just click on "Contact me" on the main page.

ISDN

[CokoBWare]

I know of a radio show in Austin, TX that is connected to the radio network located in MN through an ISDN line. It's clean, clear, and digital. I don't know the kind of equipment they use, but it is a direct digital channel between both points, and I would highly investigate this as an option. It may cost money, but it's likely worth it ($50-75/month my best guess). Check your local telecos.