When we designed SureFlow we started with a clean sheet. Other manufacturers had already started to offer their redundant streaming solutions... but we just couldn't see what was so great about them. We knew that there had to be a MUCH better way than what was being offered. What bothered us most was that the claims being made for some of these 'technologies' and the premise quoted behind the design didn't look like they were written by someone who truly understood the nature and behaviour of real world networks - let alone someone who has any understanding of live audio over IP and the Internet. Try reading it critically and it doesn't make sense. Look behind the slogans and promises - or ask a question like 'how?' or 'why?' and silence or 'it just does' is the reply.

The basic redundant streaming principle being offered elsewhere today is this: Audio is encoded and made into packets as normal. Then, at the very last stage before these are sent on the network, a duplicate copy of each packet is made and also sent. The receiving unit then, hopefully, gets at least one of the 2 copies and plays the audio from whichever it received first. Essentially the receiving device gets to switch between two identical streams. If they aren't both faulty at the same time then this is ok. There's a lot more mumbo-jumbo, but this is basically all it is.

But this assumes that you have two connections with approximately the same capacity and reliability... or that you're willing to put up with using only the lowest bitrate (quality) that both of your connections will support... and that 2 connections is sure enough. Why not use the highest bitrate that each of your connections will individually support. And why just use 2 streams/connections? SureFlow encodes each stream independently, so you can use a really fast internet connection (or other network connection) to send linear, lossless or near-lossless, or other higher bitrate audio and send a lower bitrate stream on another (for example MPEG4 AAC). And why not even run a low bitrate stream over an additional office internet connection - after all, it's already there. How about a fourth running over a 3G/4G wireless connection at as little as 14kbps? And always use the best available quality at the receiving end... that's just the start of what SureFlow offers.

You can encode your audio in up to 5 different audio types and bitrates, add FEC individually to any of these independent streams if you wish, send each stream over the same or multiple different networks (and route these via your existing routers and switches and network management) and know that STL-IP Plus will always choose the best available quality of audio, of course acting according to your chosen stream priorities, seamlessly picking and choosing and fixing right down to each individual sample of audio.

And there's a switchable safety/jitter buffer that holds up to 60 seconds of audio (set this in 1ms increments if you wish) and then even define multiple options for alternative backup audio.

SureFlow works in almost as many different scenarios as there are networks in the world! Maybe you have a private or point to point wireless network connection as your primary connection - and you want to back it up with one or more internet or other connections. That's fine. Perhaps you just want to combine internet connections - ok, well use as many as you have... maybe even add some more later... and share some of these with other uses like the office internet connection. Or do a remote/outside broadcast and combine an internet connection at the venue with a 3G/4G mobile data connection, plus maybe a third connection borrowed from a house nearby.

Have a look at the About SureFlow page of this site for more information and ideas on how to use it!