More on FreeSWITCH

I've hinted at it for some time:

I've been playing with FreeSWITCH.

Anyone who is reading this should already know what FreeSWITCH is and why someone (such as myself) would be so interested in it. I'm not going to go over all of that again; there are plenty of rave reviews all over the internet. I don't need to write another one (although I probably will some day).

Here's an update on what I've done so far:

1) FreeSWITCH support in AstLinux. Still coming along but much progress has already been made. It compiles cleanly (one more hack for sqlite) and appears to work. More testing soon but I was pleasantly surprised - the build system seems to be just as well designed as the rest of the project. They've done a great job!

2) I hate transcoding. Long, long ago I led an effort to re-record and convert all of the Asterisk prompts to various native file formats to avoid transcoding. More than two years later I'm doing it again for FreeSWITCH although this time I don't have to pay to re-record them all! Luckily they are made available in various sample rates already. I just had to update the script and do the converting. Big thanks to sox, Asterisk/res_convert and FreeSWITCH/Mod_native_file.

3) This one is barely worth mentioning but I've started (SVN branch, that's about it) working to re-implement recqual using FreeSWITCH to place calls. I've got some big plans for this. Let's see how much time I actually have to work on it. Don't expect much progress anytime soon.

4) Various production and consulting projects. Obviously.

As always, expect more to come!

Introducing Recqual

I've been waiting to talk about this one for a while.

Several months ago Star2Star was having problems with one of our upstream SIP carriers. We were starting to notice a large increase in the number of one way audio calls our customers were reporting.

When most people think of one way calls their first reaction is to blame SIP. Must be NAT! Must be a firewall! SIP sucks! Etc, etc.

I knew that wasn't the case. I just had to prove it.

I was convinced the problem wasn't SIP/UDP/IP related at all. We had multiple pcaps where we were sending RTP to the appropriate gateway. It just wasn't getting to the PSTN. Where was it going? When was this happening? Which gateways (out of hundreds) were the most problematic? We needed to know and we needed to know quickly.

I came up with and "wrote" recqual over a couple of days. After a few runs we were noticing patterns with problematic RTP endpoint IP addresses. Long story short, once these were identified we worked with the carrier to replace various bits of equipment (DSPs, line cards, etc). The one way audio problem has largely disappeared and we continue to run recqual. If this starts happening again we should know /BEFORE/ our customers do.

Of course I'm using Asterisk to place the calls. The best part of using Asterisk is it's multi-protocol flexibility. You should be able to test just about any combination of voice technologies - G.279a, G711, GSM, SIP, IAX, PRI, FXO, FXO, gtalk/jabber/jingle, skype, etc. The possibilities boggle the mind.

I've just been too busy to get it together and release this to the community - until now.

Tarball with instructions here.

Questions? Comments? Suggestions? Drop me a line.