By Aaron Huslage
The world of open source VoIP platforms has been dominated by the Asterisk PBX and the products that surround it. While no one has quantified the market share numbers for this beast, it has more than likely over 95 percent of the open source share and some significant portion of IP PBXs installed around the world. As Om wrote in late 2007, Asterisk has been downloaded over 1 million times. The case for using open source for critical calling infrastructure has been made by the success of this single product. The ecosystem surrounding Asterisk is huge and continues to grow apace, and now another player is poised to give Asterisk a run for its money.
Announced at the Emerging Telephony conference in 2006, Freeswitch has been quietly growing in both popularity and stability not as a direct competitor to Asterisk, but as a fully featured carrier-grade Softswitch. I joined in on a conference call with the team developing Freeswitch to find out more as they near their 1.0 release.
Historically, a softswitch is designed to be the stable core of a network, first and foremost. It is meant to replace big-iron switches like the DMS-10 from Nortel with commodity hardware running off the shelf operating systems. A PBX is meant to be the phone system you use at your desk in your office. It provides voicemail and extensions for everyone, but it doesn't do any major call routing or application serving.
Freeswitch tends to blur this line a bit, but as lead developer Tony Minessale explained, "We can do PBX stuff, but that's more of a given. The more advanced stuff we do kind of requires a lot of that code, so there isn't a big disadvantage to using it like a PBX. But we really see it implemented in a bigger manner." Freeswitch is more a welcome addition to the world of open source VoIP than a direct competitor to Asterisk.
"Freeswitch is a perfect platform for CLECs to deploy innovative new applications like conferencing and voice mail." said Ken Rice, developer and user of the system. "We took out a Tier-2 carrier's $1 million switch because a couple of our Freeswitch boxes provided too much call volume at one time." This is an astounding statement of the performance of this system. The group claims that the software has been tested for 100 hours to over 10 million calls without crashing. "One of our switches has 2500 call legs and it's using a single Dell 1950 and the memory and processor utilization is only about 10%. It's doing 80-100 call setups per second," said Rice.
The team is ready to roll with their final 1.0 release. Developer Mike Jerris told me that they have "spent 2 to 3 months working on stabilization between live traffic implementations. We think we've found all the real-life edge cases at this point." The code has reached a maturity level that the community and the developers feel is ready for the big milestone of a 1.0 release on May 28. The current release is Release Candidate 2 and is available now from the Freeswitch web site.
Are you familiar with Freeswitch and/or Asterisk?
Comments
Add CommentBy Contact Center Comanche on Apr. 14, 2008
We've been using FreeSwitch for about 3 months and are pounding a lot of cheap SIP trunking minutes through AireSpring. We've saved a lot of money since we started. However, it was a bit difficult to get started up at first. I'm sure that as the FreeSwitch development community gets ramped up, this will get easier and easier.
By an anonymous user on Apr. 15, 2008
I used Asterisk a year or so ago. I was hoping to get something that I could run on a machine, fire up a soft phone, connect it and have it be pbx-ready. Asterisk was not there at that point. Not sure if it is now either - I need to look in to a virtual appliance, perhaps, since the setup was cumbersome too for a relatively tech-savvy user, but not a pbx systems expert....
By Michael S Collins on Apr. 15, 2008
I have used both Asterisk and FreeSWITCH. In fact, I'm helping the FS devs test the TDM stuff (OpenZAP) by using an Asterisk system as a make-shift telco. Asterisk is a good PBX, but it does have its quirks and limitations. I suspect that we'll start seeing more cases of mixed environments, e.g. FS as a PSTN gateway for a SIPx PBX (a user is doing this currently) or FS as a "super router" for VoIP calls that are generated by Asterisk or other IPPBX systems. (This too is in actual production.)
Gotta love OSS - just download them both and take them for a test drive. That's what I did and I chose FS as the project to which I would donate time/money/energy as well as to use in production.
-MC
By Randy Claines on Jul. 19, 2008
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